diff options
| author | Android Chromium Automerger <chromium-automerger@android> | 2014-07-29 12:29:13 +0000 |
|---|---|---|
| committer | Android Chromium Automerger <chromium-automerger@android> | 2014-07-29 12:29:13 +0000 |
| commit | fa5085428e6672f7f9e88a652e3c4a7f9c93d116 (patch) | |
| tree | ef453f1196cff48f99c0b0f45a95c11a4fd49b97 /modules | |
| parent | f3d27028ce9a280c02c286f9b98adf1e476c03d2 (diff) | |
| parent | 4a1b3e3a69d349b0d3e91f607f24e02d8b975688 (diff) | |
| download | webrtc-fa5085428e6672f7f9e88a652e3c4a7f9c93d116.tar.gz | |
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 4a1b3e3a69d349b0d3e91f607f24e02d8b975688
This commit was generated by merge_from_chromium.py.
Change-Id: Iada7abd78f123301a98db982a6272cd9487de72f
Diffstat (limited to 'modules')
89 files changed, 3405 insertions, 2178 deletions
diff --git a/modules/audio_coding/codecs/isac/fix/source/codec.h b/modules/audio_coding/codecs/isac/fix/source/codec.h index 2f649324..a38c6e56 100644 --- a/modules/audio_coding/codecs/isac/fix/source/codec.h +++ b/modules/audio_coding/codecs/isac/fix/source/codec.h @@ -101,6 +101,16 @@ void WebRtcIsacfix_Spec2TimeNeon(int16_t* inreQ7, int32_t* outre2Q16); #endif +#if defined(MIPS32_LE) +void WebRtcIsacfix_Time2SpecMIPS(int16_t* inre1Q9, + int16_t* inre2Q9, + int16_t* outre, + int16_t* outim); +void WebRtcIsacfix_Spec2TimeMIPS(int16_t* inreQ7, + int16_t* inimQ7, + int32_t* outre1Q16, + int32_t* outre2Q16); +#endif /* filterbank functions */ diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h b/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h index 3fefc1a5..7a5f7462 100644 --- a/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h +++ b/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h @@ -23,10 +23,23 @@ extern "C" { * coefficient: Input. * state: Input/output, filter state, in Q4. */ -void WebRtcIsacfix_HighpassFilterFixDec32(int16_t *io, - int16_t len, - const int16_t *coefficient, - int32_t *state); +typedef void (*HighpassFilterFixDec32)(int16_t* io, + int16_t len, + const int16_t* coefficient, + int32_t* state); +extern HighpassFilterFixDec32 WebRtcIsacfix_HighpassFilterFixDec32; + +void WebRtcIsacfix_HighpassFilterFixDec32C(int16_t* io, + int16_t len, + const int16_t* coefficient, + int32_t* state); + +#if defined(MIPS_DSP_R1_LE) +void WebRtcIsacfix_HighpassFilterFixDec32MIPS(int16_t* io, + int16_t len, + const int16_t* coefficient, + int32_t* state); +#endif typedef void (*AllpassFilter2FixDec16)( int16_t *data_ch1, // Input and output in channel 1, in Q0 diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbanks.c b/modules/audio_coding/codecs/isac/fix/source/filterbanks.c index 64557e13..1928a7cb 100644 --- a/modules/audio_coding/codecs/isac/fix/source/filterbanks.c +++ b/modules/audio_coding/codecs/isac/fix/source/filterbanks.c @@ -86,10 +86,13 @@ void WebRtcIsacfix_AllpassFilter2FixDec16C( filter_state_ch2[1] = state1_ch2; } -void WebRtcIsacfix_HighpassFilterFixDec32(int16_t *io, - int16_t len, - const int16_t *coefficient, - int32_t *state) +// Declare a function pointer. +HighpassFilterFixDec32 WebRtcIsacfix_HighpassFilterFixDec32; + +void WebRtcIsacfix_HighpassFilterFixDec32C(int16_t *io, + int16_t len, + const int16_t *coefficient, + int32_t *state) { int k; int32_t a1 = 0, b1 = 0, c = 0, in = 0; diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbanks_mips.c b/modules/audio_coding/codecs/isac/fix/source/filterbanks_mips.c index 1887745b..4dd70cf6 100644 --- a/modules/audio_coding/codecs/isac/fix/source/filterbanks_mips.c +++ b/modules/audio_coding/codecs/isac/fix/source/filterbanks_mips.c @@ -10,26 +10,26 @@ #include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h" -// WebRtcIsacfix_AllpassFilter2FixDec16 function optimized for MIPSDSP platform -// Bit-exact with WebRtcIsacfix_AllpassFilter2FixDec16C from filterbanks.c +// WebRtcIsacfix_AllpassFilter2FixDec16 function optimized for MIPSDSP platform. +// Bit-exact with WebRtcIsacfix_AllpassFilter2FixDec16C from filterbanks.c. void WebRtcIsacfix_AllpassFilter2FixDec16MIPS( - int16_t *data_ch1, // Input and output in channel 1, in Q0 - int16_t *data_ch2, // Input and output in channel 2, in Q0 - const int16_t *factor_ch1, // Scaling factor for channel 1, in Q15 - const int16_t *factor_ch2, // Scaling factor for channel 2, in Q15 - const int length, // Length of the data buffers - int32_t *filter_state_ch1, // Filter state for channel 1, in Q16 - int32_t *filter_state_ch2) { // Filter state for channel 2, in Q16 + int16_t* data_ch1, // Input and output in channel 1, in Q0. + int16_t* data_ch2, // Input and output in channel 2, in Q0. + const int16_t* factor_ch1, // Scaling factor for channel 1, in Q15. + const int16_t* factor_ch2, // Scaling factor for channel 2, in Q15. + const int length, // Length of the data buffers. + int32_t* filter_state_ch1, // Filter state for channel 1, in Q16. + int32_t* filter_state_ch2) { // Filter state for channel 2, in Q16. - int32_t st0_ch1, st1_ch1; // channel1 state variables - int32_t st0_ch2, st1_ch2; // channel2 state variables - int32_t f_ch10, f_ch11, f_ch20, f_ch21; // factor variables - int32_t r0, r1, r2, r3, r4, r5; // temporary ragister variables + int32_t st0_ch1, st1_ch1; // channel1 state variables. + int32_t st0_ch2, st1_ch2; // channel2 state variables. + int32_t f_ch10, f_ch11, f_ch20, f_ch21; // factor variables. + int32_t r0, r1, r2, r3, r4, r5; // temporary register variables. __asm __volatile ( ".set push \n\t" ".set noreorder \n\t" - // Load all the state and factor variables + // Load all the state and factor variables. "lh %[f_ch10], 0(%[factor_ch1]) \n\t" "lh %[f_ch20], 0(%[factor_ch2]) \n\t" "lh %[f_ch11], 2(%[factor_ch1]) \n\t" @@ -38,7 +38,7 @@ void WebRtcIsacfix_AllpassFilter2FixDec16MIPS( "lw %[st1_ch1], 4(%[filter_state_ch1]) \n\t" "lw %[st0_ch2], 0(%[filter_state_ch2]) \n\t" "lw %[st1_ch2], 4(%[filter_state_ch2]) \n\t" - // Allpass filtering loop + // Allpass filtering loop. "1: \n\t" "lh %[r0], 0(%[data_ch1]) \n\t" "lh %[r1], 0(%[data_ch2]) \n\t" @@ -80,7 +80,7 @@ void WebRtcIsacfix_AllpassFilter2FixDec16MIPS( "subq_s.w %[st1_ch2], %[r3], %[r1] \n\t" "bgtz %[length], 1b \n\t" " addiu %[data_ch2], %[data_ch2], 2 \n\t" - // Store channel states + // Store channel states. "sw %[st0_ch1], 0(%[filter_state_ch1]) \n\t" "sw %[st1_ch1], 4(%[filter_state_ch1]) \n\t" "sw %[st0_ch2], 0(%[filter_state_ch2]) \n\t" @@ -100,3 +100,143 @@ void WebRtcIsacfix_AllpassFilter2FixDec16MIPS( : "memory", "hi", "lo" ); } + +// WebRtcIsacfix_HighpassFilterFixDec32 function optimized for MIPSDSP platform. +// Bit-exact with WebRtcIsacfix_HighpassFilterFixDec32C from filterbanks.c. +void WebRtcIsacfix_HighpassFilterFixDec32MIPS(int16_t* io, + int16_t len, + const int16_t* coefficient, + int32_t* state) { + int k; + int32_t a1, a2, b1, b2, in; + int32_t state0 = state[0]; + int32_t state1 = state[1]; + + int32_t c0, c1, c2, c3; + int32_t c4, c5, c6, c7; + int32_t state0_lo, state0_hi; + int32_t state1_lo, state1_hi; + int32_t t0, t1, t2, t3, t4, t5; + + __asm __volatile ( + "lh %[c0], 0(%[coeff_ptr]) \n\t" + "lh %[c1], 2(%[coeff_ptr]) \n\t" + "lh %[c2], 4(%[coeff_ptr]) \n\t" + "lh %[c3], 6(%[coeff_ptr]) \n\t" + "sra %[state0_hi], %[state0], 16 \n\t" + "sra %[state1_hi], %[state1], 16 \n\t" + "andi %[state0_lo], %[state0], 0xFFFF \n\t" + "andi %[state1_lo], %[state1], 0xFFFF \n\t" + "lh %[c4], 8(%[coeff_ptr]) \n\t" + "lh %[c5], 10(%[coeff_ptr]) \n\t" + "lh %[c6], 12(%[coeff_ptr]) \n\t" + "lh %[c7], 14(%[coeff_ptr]) \n\t" + "sra %[state0_lo], %[state0_lo], 1 \n\t" + "sra %[state1_lo], %[state1_lo], 1 \n\t" + : [c0] "=&r" (c0), [c1] "=&r" (c1), [c2] "=&r" (c2), [c3] "=&r" (c3), + [c4] "=&r" (c4), [c5] "=&r" (c5), [c6] "=&r" (c6), [c7] "=&r" (c7), + [state0_hi] "=&r" (state0_hi), [state0_lo] "=&r" (state0_lo), + [state1_hi] "=&r" (state1_hi), [state1_lo] "=&r" (state1_lo) + : [coeff_ptr] "r" (coefficient), [state0] "r" (state0), + [state1] "r" (state1) + : "memory" + ); + + for (k = 0; k < len; k++) { + in = (int32_t)io[k]; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "mul %[t2], %[c4], %[state0_lo] \n\t" + "mul %[t0], %[c5], %[state0_lo] \n\t" + "mul %[t1], %[c4], %[state0_hi] \n\t" + "mul %[a1], %[c5], %[state0_hi] \n\t" + "mul %[t5], %[c6], %[state1_lo] \n\t" + "mul %[t3], %[c7], %[state1_lo] \n\t" + "mul %[t4], %[c6], %[state1_hi] \n\t" + "mul %[b1], %[c7], %[state1_hi] \n\t" + "shra_r.w %[t2], %[t2], 15 \n\t" + "shra_r.w %[t0], %[t0], 15 \n\t" + "addu %[t1], %[t1], %[t2] \n\t" + "addu %[a1], %[a1], %[t0] \n\t" + "sra %[t1], %[t1], 16 \n\t" + "addu %[a1], %[a1], %[t1] \n\t" + "shra_r.w %[t5], %[t5], 15 \n\t" + "shra_r.w %[t3], %[t3], 15 \n\t" + "addu %[t4], %[t4], %[t5] \n\t" + "addu %[b1], %[b1], %[t3] \n\t" + "sra %[t4], %[t4], 16 \n\t" + "addu %[b1], %[b1], %[t4] \n\t" + "mul %[t2], %[c0], %[state0_lo] \n\t" + "mul %[t0], %[c1], %[state0_lo] \n\t" + "mul %[t1], %[c0], %[state0_hi] \n\t" + "mul %[a2], %[c1], %[state0_hi] \n\t" + "mul %[t5], %[c2], %[state1_lo] \n\t" + "mul %[t3], %[c3], %[state1_lo] \n\t" + "mul %[t4], %[c2], %[state1_hi] \n\t" + "mul %[b2], %[c3], %[state1_hi] \n\t" + "shra_r.w %[t2], %[t2], 15 \n\t" + "shra_r.w %[t0], %[t0], 15 \n\t" + "addu %[t1], %[t1], %[t2] \n\t" + "addu %[a2], %[a2], %[t0] \n\t" + "sra %[t1], %[t1], 16 \n\t" + "addu %[a2], %[a2], %[t1] \n\t" + "shra_r.w %[t5], %[t5], 15 \n\t" + "shra_r.w %[t3], %[t3], 15 \n\t" + "addu %[t4], %[t4], %[t5] \n\t" + "addu %[b2], %[b2], %[t3] \n\t" + "sra %[t4], %[t4], 16 \n\t" + "addu %[b2], %[b2], %[t4] \n\t" + "addu %[a1], %[a1], %[b1] \n\t" + "sra %[a1], %[a1], 7 \n\t" + "addu %[a1], %[a1], %[in] \n\t" + "sll %[t0], %[in], 2 \n\t" + "addu %[a2], %[a2], %[b2] \n\t" + "subu %[t0], %[t0], %[a2] \n\t" + "shll_s.w %[a1], %[a1], 16 \n\t" + "shll_s.w %[t0], %[t0], 2 \n\t" + "sra %[a1], %[a1], 16 \n\t" + "addu %[state1_hi], %[state0_hi], $0 \n\t" + "addu %[state1_lo], %[state0_lo], $0 \n\t" + "sra %[state0_hi], %[t0], 16 \n\t" + "andi %[state0_lo], %[t0], 0xFFFF \n\t" + "sra %[state0_lo], %[state0_lo], 1 \n\t" + ".set pop \n\t" + : [a1] "=&r" (a1), [b1] "=&r" (b1), [a2] "=&r" (a2), [b2] "=&r" (b2), + [state0_hi] "+r" (state0_hi), [state0_lo] "+r" (state0_lo), + [state1_hi] "+r" (state1_hi), [state1_lo] "+r" (state1_lo), + [t0] "=&r" (t0), [t1] "=&r" (t1), [t2] "=&r" (t2), + [t3] "=&r" (t3), [t4] "=&r" (t4), [t5] "=&r" (t5) + : [c0] "r" (c0), [c1] "r" (c1), [c2] "r" (c2), [c3] "r" (c3), + [c4] "r" (c4), [c5] "r" (c5), [c6] "r" (c6), [c7] "r" (c7), + [in] "r" (in) + : "hi", "lo" + ); + io[k] = (int16_t)a1; + } + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" +#if !defined(MIPS_DSP_R2_LE) + "sll %[state0_hi], %[state0_hi], 16 \n\t" + "sll %[state0_lo], %[state0_lo], 1 \n\t" + "sll %[state1_hi], %[state1_hi], 16 \n\t" + "sll %[state1_lo], %[state1_lo], 1 \n\t" + "or %[state0_hi], %[state0_hi], %[state0_lo] \n\t" + "or %[state1_hi], %[state1_hi], %[state1_lo] \n\t" +#else + "sll %[state0_lo], %[state0_lo], 1 \n\t" + "sll %[state1_lo], %[state1_lo], 1 \n\t" + "precr_sra.ph.w %[state0_hi], %[state0_lo], 0 \n\t" + "precr_sra.ph.w %[state1_hi], %[state1_lo], 0 \n\t" +#endif + "sw %[state0_hi], 0(%[state]) \n\t" + "sw %[state1_hi], 4(%[state]) \n\t" + ".set pop \n\t" + : [state0_hi] "+r" (state0_hi), [state0_lo] "+r" (state0_lo), + [state1_hi] "+r" (state1_hi), [state1_lo] "+r" (state1_lo) + : [state] "r" (state) + : "memory" + ); +} diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc b/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc index d7484277..d15318a7 100644 --- a/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc +++ b/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc @@ -86,6 +86,13 @@ TEST_F(FilterBanksTest, HighpassFilterFixDec32Test) { -1280, -8554, -14496, -7561, -23541, -27263, -30560, -32768, -3441, -32768, 25203, -27550, 22419}; #endif + HighpassFilterFixDec32 WebRtcIsacfix_HighpassFilterFixDec32; +#if defined(MIPS_DSP_R1_LE) + WebRtcIsacfix_HighpassFilterFixDec32 = + WebRtcIsacfix_HighpassFilterFixDec32MIPS; +#else + WebRtcIsacfix_HighpassFilterFixDec32 = WebRtcIsacfix_HighpassFilterFixDec32C; +#endif for (int i = 0; i < kSamples; i++) { in[i] = WEBRTC_SPL_WORD32_MAX / (i + 1); diff --git a/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/modules/audio_coding/codecs/isac/fix/source/isacfix.c index 76359080..887a7ba2 100644 --- a/modules/audio_coding/codecs/isac/fix/source/isacfix.c +++ b/modules/audio_coding/codecs/isac/fix/source/isacfix.c @@ -209,9 +209,17 @@ static void WebRtcIsacfix_InitNeon(void) { static void WebRtcIsacfix_InitMIPS(void) { WebRtcIsacfix_AutocorrFix = WebRtcIsacfix_AutocorrMIPS; WebRtcIsacfix_FilterMaLoopFix = WebRtcIsacfix_FilterMaLoopMIPS; + WebRtcIsacfix_Spec2Time = WebRtcIsacfix_Spec2TimeMIPS; + WebRtcIsacfix_Time2Spec = WebRtcIsacfix_Time2SpecMIPS; #if defined(MIPS_DSP_R1_LE) WebRtcIsacfix_AllpassFilter2FixDec16 = WebRtcIsacfix_AllpassFilter2FixDec16MIPS; + WebRtcIsacfix_HighpassFilterFixDec32 = + WebRtcIsacfix_HighpassFilterFixDec32MIPS; +#endif +#if defined(MIPS_DSP_R2_LE) + WebRtcIsacfix_CalculateResidualEnergy = + WebRtcIsacfix_CalculateResidualEnergyMIPS; #endif } #endif @@ -300,10 +308,11 @@ int16_t WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst, WebRtcIsacfix_CalculateResidualEnergy = WebRtcIsacfix_CalculateResidualEnergyC; WebRtcIsacfix_AllpassFilter2FixDec16 = WebRtcIsacfix_AllpassFilter2FixDec16C; + WebRtcIsacfix_HighpassFilterFixDec32 = WebRtcIsacfix_HighpassFilterFixDec32C; WebRtcIsacfix_Time2Spec = WebRtcIsacfix_Time2SpecC; WebRtcIsacfix_Spec2Time = WebRtcIsacfix_Spec2TimeC; WebRtcIsacfix_MatrixProduct1 = WebRtcIsacfix_MatrixProduct1C; - WebRtcIsacfix_MatrixProduct2 = WebRtcIsacfix_MatrixProduct2C ; + WebRtcIsacfix_MatrixProduct2 = WebRtcIsacfix_MatrixProduct2C; #ifdef WEBRTC_DETECT_ARM_NEON if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) { diff --git a/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi b/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi index a18a803d..e5aade65 100644 --- a/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi +++ b/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi @@ -47,12 +47,14 @@ 'lpc_masking_model.c', 'lpc_tables.c', 'pitch_estimator.c', + 'pitch_estimator_c.c', 'pitch_filter.c', 'pitch_filter_c.c', 'pitch_gain_tables.c', 'pitch_lag_tables.c', 'spectrum_ar_model_tables.c', 'transform.c', + 'transform_tables.c', 'arith_routins.h', 'bandwidth_estimator.h', 'codec.h', @@ -89,9 +91,12 @@ 'sources': [ 'filters_mips.c', 'lattice_mips.c', + 'pitch_estimator_mips.c', + 'transform_mips.c', ], 'sources!': [ 'lattice_c.c', + 'pitch_estimator_c.c', ], 'conditions': [ ['mips_dsp_rev>0', { @@ -101,6 +106,7 @@ }], ['mips_dsp_rev>1', { 'sources': [ + 'lpc_masking_model_mips.c', 'pitch_filter_mips.c', ], 'sources!': [ diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h index 72e0cfc4..1270c142 100644 --- a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h +++ b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h @@ -62,6 +62,15 @@ int32_t WebRtcIsacfix_CalculateResidualEnergyNeon(int lpc_order, int* q_val_residual_energy); #endif +#if defined(MIPS_DSP_R2_LE) +int32_t WebRtcIsacfix_CalculateResidualEnergyMIPS(int lpc_order, + int32_t q_val_corr, + int q_val_polynomial, + int16_t* a_polynomial, + int32_t* corr_coeffs, + int* q_val_residual_energy); +#endif + #ifdef __cplusplus } /* extern "C" */ #endif diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_mips.c b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_mips.c new file mode 100644 index 00000000..55602b97 --- /dev/null +++ b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_mips.c @@ -0,0 +1,237 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h" + +// MIPS DSPR2 optimization for function WebRtcIsacfix_CalculateResidualEnergy +// Bit-exact with WebRtcIsacfix_CalculateResidualEnergyC from file +// lpc_masking_model.c +int32_t WebRtcIsacfix_CalculateResidualEnergyMIPS(int lpc_order, + int32_t q_val_corr, + int q_val_polynomial, + int16_t* a_polynomial, + int32_t* corr_coeffs, + int* q_val_residual_energy) { + + int i = 0, j = 0; + int shift_internal = 0, shift_norm = 0; + int32_t tmp32 = 0, word32_high = 0, word32_low = 0, residual_energy = 0; + int32_t tmp_corr_c = corr_coeffs[0]; + int16_t* tmp_a_poly = &a_polynomial[0]; + int32_t sum64_hi = 0; + int32_t sum64_lo = 0; + + for (j = 0; j <= lpc_order; j++) { + // For the case of i == 0: + // residual_energy += + // a_polynomial[j] * corr_coeffs[i] * a_polynomial[j - i]; + + int32_t tmp2, tmp3; + int16_t sign_1; + int16_t sign_2; + int16_t sign_3; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "lh %[tmp2], 0(%[tmp_a_poly]) \n\t" + "mul %[tmp32], %[tmp2], %[tmp2] \n\t" + "addiu %[tmp_a_poly], %[tmp_a_poly], 2 \n\t" + "sra %[sign_2], %[sum64_hi], 31 \n\t" + "mult $ac0, %[tmp32], %[tmp_corr_c] \n\t" + "shilov $ac0, %[shift_internal] \n\t" + "mfhi %[tmp2], $ac0 \n\t" + "mflo %[tmp3], $ac0 \n\t" + "sra %[sign_1], %[tmp2], 31 \n\t" + "xor %[sign_3], %[sign_1], %[sign_2] \n\t" + ".set pop \n\t" + : [tmp2] "=&r" (tmp2), [tmp3] "=&r" (tmp3), [tmp32] "=&r" (tmp32), + [tmp_a_poly] "+r" (tmp_a_poly), [sign_1] "=&r" (sign_1), + [sign_3] "=&r" (sign_3), [sign_2] "=&r" (sign_2), + [sum64_hi] "+r" (sum64_hi), [sum64_lo] "+r" (sum64_lo) + : [tmp_corr_c] "r" (tmp_corr_c), [shift_internal] "r" (shift_internal) + : "hi", "lo", "memory" + ); + + if (sign_3 != 0) { + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addsc %[sum64_lo], %[sum64_lo], %[tmp3] \n\t" + "addwc %[sum64_hi], %[sum64_hi], %[tmp2] \n\t" + ".set pop \n\t" + : [sum64_hi] "+r" (sum64_hi), [sum64_lo] "+r" (sum64_lo) + : [tmp2] "r" (tmp2), [tmp3] "r" (tmp3) + : "hi", "lo", "memory" + ); + } else { + if (((!(sign_1 || sign_2)) && (0x7FFFFFFF - sum64_hi < tmp2)) || + ((sign_1 && sign_2) && (sum64_hi + tmp2 > 0))) { + // Shift right for overflow. + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[shift_internal], %[shift_internal], 1 \n\t" + "prepend %[sum64_lo], %[sum64_hi], 1 \n\t" + "sra %[sum64_hi], %[sum64_hi], 1 \n\t" + "prepend %[tmp3], %[tmp2], 1 \n\t" + "sra %[tmp2], %[tmp2], 1 \n\t" + "addsc %[sum64_lo], %[sum64_lo], %[tmp3] \n\t" + "addwc %[sum64_hi], %[sum64_hi], %[tmp2] \n\t" + ".set pop \n\t" + : [tmp2] "+r" (tmp2), [tmp3] "+r" (tmp3), + [shift_internal] "+r" (shift_internal), + [sum64_hi] "+r" (sum64_hi), [sum64_lo] "+r" (sum64_lo) + : + : "hi", "lo", "memory" + ); + } else { + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addsc %[sum64_lo], %[sum64_lo], %[tmp3] \n\t" + "addwc %[sum64_hi], %[sum64_hi], %[tmp2] \n\t" + ".set pop \n\t" + : [sum64_hi] "+r" (sum64_hi), [sum64_lo] "+r" (sum64_lo) + : [tmp2] "r" (tmp2), [tmp3] "r" (tmp3) + : "hi", "lo", "memory" + ); + } + } + } + + for (i = 1; i <= lpc_order; i++) { + tmp_corr_c = corr_coeffs[i]; + int16_t* tmp_a_poly_j = &a_polynomial[i]; + int16_t* tmp_a_poly_j_i = &a_polynomial[0]; + for (j = i; j <= lpc_order; j++) { + // For the case of i = 1 .. lpc_order: + // residual_energy += + // a_polynomial[j] * corr_coeffs[i] * a_polynomial[j - i] * 2; + + int32_t tmp2, tmp3; + int16_t sign_1; + int16_t sign_2; + int16_t sign_3; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "lh %[tmp3], 0(%[tmp_a_poly_j]) \n\t" + "lh %[tmp2], 0(%[tmp_a_poly_j_i]) \n\t" + "addiu %[tmp_a_poly_j], %[tmp_a_poly_j], 2 \n\t" + "addiu %[tmp_a_poly_j_i], %[tmp_a_poly_j_i], 2 \n\t" + "mul %[tmp32], %[tmp3], %[tmp2] \n\t" + "sll %[tmp32], %[tmp32], 1 \n\t" + "mult $ac0, %[tmp32], %[tmp_corr_c] \n\t" + "shilov $ac0, %[shift_internal] \n\t" + "mfhi %[tmp2], $ac0 \n\t" + "mflo %[tmp3], $ac0 \n\t" + "sra %[sign_1], %[tmp2], 31 \n\t" + "sra %[sign_2], %[sum64_hi], 31 \n\t" + "xor %[sign_3], %[sign_1], %[sign_2] \n\t" + ".set pop \n\t" + : [tmp2] "=&r" (tmp2), [tmp3] "=&r" (tmp3), [tmp32] "=&r" (tmp32), + [tmp_a_poly_j] "+r" (tmp_a_poly_j), [sign_1] "=&r" (sign_1), + [tmp_a_poly_j_i] "+r" (tmp_a_poly_j_i), [sign_2] "=&r" (sign_2), + [sign_3] "=&r" (sign_3), [sum64_hi] "+r" (sum64_hi), + [sum64_lo] "+r" (sum64_lo) + : [tmp_corr_c] "r" (tmp_corr_c), [shift_internal] "r" (shift_internal) + : "hi", "lo", "memory" + ); + if (sign_3 != 0) { + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addsc %[sum64_lo], %[sum64_lo], %[tmp3] \n\t" + "addwc %[sum64_hi], %[sum64_hi], %[tmp2] \n\t" + ".set pop \n\t" + : [tmp2] "+r" (tmp2), [tmp3] "+r" (tmp3), [sum64_hi] "+r" (sum64_hi), + [sum64_lo] "+r" (sum64_lo) + : + :"memory" + ); + } else { + // Test overflow and sum the result. + if (((!(sign_1 || sign_2)) && (0x7FFFFFFF - sum64_hi < tmp2)) || + ((sign_1 && sign_2) && (sum64_hi + tmp2 > 0))) { + // Shift right for overflow. + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[shift_internal], %[shift_internal], 1 \n\t" + "prepend %[sum64_lo], %[sum64_hi], 1 \n\t" + "sra %[sum64_hi], %[sum64_hi], 1 \n\t" + "prepend %[tmp3], %[tmp2], 1 \n\t" + "sra %[tmp2], %[tmp2], 1 \n\t" + "addsc %[sum64_lo], %[sum64_lo], %[tmp3] \n\t" + "addwc %[sum64_hi], %[sum64_hi], %[tmp2] \n\t" + ".set pop \n\t" + : [tmp2] "+r" (tmp2), [tmp3] "+r" (tmp3), + [shift_internal] "+r" (shift_internal), + [sum64_hi] "+r" (sum64_hi), [sum64_lo] "+r" (sum64_lo) + : + : "hi", "lo", "memory" + ); + } else { + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addsc %[sum64_lo], %[sum64_lo], %[tmp3] \n\t" + "addwc %[sum64_hi], %[sum64_hi], %[tmp2] \n\t" + ".set pop \n\t" + : [tmp2] "+r" (tmp2), [tmp3] "+r" (tmp3), + [sum64_hi] "+r" (sum64_hi), [sum64_lo] "+r" (sum64_lo) + : + : "hi", "lo", "memory" + ); + } + } + } + } + word32_high = sum64_hi; + word32_low = sum64_lo; + + // Calculate the value of shifting (shift_norm) for the 64-bit sum. + if (word32_high != 0) { + shift_norm = 32 - WebRtcSpl_NormW32(word32_high); + int tmp1; + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "srl %[residual_energy], %[sum64_lo], %[shift_norm] \n\t" + "li %[tmp1], 32 \n\t" + "subu %[tmp1], %[tmp1], %[shift_norm] \n\t" + "sll %[tmp1], %[sum64_hi], %[tmp1] \n\t" + "or %[residual_energy], %[residual_energy], %[tmp1] \n\t" + ".set pop \n\t" + : [residual_energy] "=&r" (residual_energy), [tmp1]"=&r"(tmp1), + [sum64_hi] "+r" (sum64_hi), [sum64_lo] "+r" (sum64_lo) + : [shift_norm] "r" (shift_norm) + : "memory" + ); + } else { + if ((word32_low & 0x80000000) != 0) { + shift_norm = 1; + residual_energy = (uint32_t)word32_low >> 1; + } else { + shift_norm = WebRtcSpl_NormW32(word32_low); + residual_energy = word32_low << shift_norm; + shift_norm = -shift_norm; + } + } + + // Q(q_val_polynomial * 2) * Q(q_val_corr) >> shift_internal >> shift_norm + // = Q(q_val_corr - shift_internal - shift_norm + q_val_polynomial * 2) + *q_val_residual_energy = + q_val_corr - shift_internal - shift_norm + q_val_polynomial * 2; + + return residual_energy; +} diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c index 9c4e5875..426b2cf4 100644 --- a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c +++ b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c @@ -29,7 +29,7 @@ static const int16_t kACoefQ12[3] = { -static __inline int32_t Log2Q8( uint32_t x ) { +__inline int32_t WebRtcIsacfix_Log2Q8( uint32_t x ) { int32_t zeros, lg2; int16_t frac; @@ -153,109 +153,7 @@ static void FindFour32(int32_t *in, int16_t length, int16_t *bestind) -static void PCorr2Q32(const int16_t *in, int32_t *logcorQ8) -{ - int16_t scaling,n,k; - int32_t ysum32,csum32, lys, lcs; - int32_t oneQ8; - - - const int16_t *x, *inptr; - - oneQ8 = WEBRTC_SPL_LSHIFT_W32((int32_t)1, 8); // 1.00 in Q8 - - x = in + PITCH_MAX_LAG/2 + 2; - scaling = WebRtcSpl_GetScalingSquare ((int16_t *) in, PITCH_CORR_LEN2, PITCH_CORR_LEN2); - ysum32 = 1; - csum32 = 0; - x = in + PITCH_MAX_LAG/2 + 2; - for (n = 0; n < PITCH_CORR_LEN2; n++) { - ysum32 += WEBRTC_SPL_MUL_16_16_RSFT( (int16_t) in[n],(int16_t) in[n], scaling); // Q0 - csum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t) x[n],(int16_t) in[n], scaling); // Q0 - } - - logcorQ8 += PITCH_LAG_SPAN2 - 1; - - lys=Log2Q8((uint32_t) ysum32); // Q8 - lys=WEBRTC_SPL_RSHIFT_W32(lys, 1); //sqrt(ysum); - - if (csum32>0) { - - lcs=Log2Q8((uint32_t) csum32); // 2log(csum) in Q8 - - if (lcs>(lys + oneQ8) ){ // csum/sqrt(ysum) > 2 in Q8 - *logcorQ8 = lcs - lys; // log2(csum/sqrt(ysum)) - } else { - *logcorQ8 = oneQ8; // 1.00 - } - - } else { - *logcorQ8 = 0; - } - - - for (k = 1; k < PITCH_LAG_SPAN2; k++) { - inptr = &in[k]; - ysum32 -= WEBRTC_SPL_MUL_16_16_RSFT( (int16_t) in[k-1],(int16_t) in[k-1], scaling); - ysum32 += WEBRTC_SPL_MUL_16_16_RSFT( (int16_t) in[PITCH_CORR_LEN2 + k - 1],(int16_t) in[PITCH_CORR_LEN2 + k - 1], scaling); - -#ifdef WEBRTC_ARCH_ARM_NEON - { - int32_t vbuff[4]; - int32x4_t int_32x4_sum = vmovq_n_s32(0); - // Can't shift a Neon register to right with a non-constant shift value. - int32x4_t int_32x4_scale = vdupq_n_s32(-scaling); - // Assert a codition used in loop unrolling at compile-time. - COMPILE_ASSERT(PITCH_CORR_LEN2 %4 == 0); - - for (n = 0; n < PITCH_CORR_LEN2; n += 4) { - int16x4_t int_16x4_x = vld1_s16(&x[n]); - int16x4_t int_16x4_in = vld1_s16(&inptr[n]); - int32x4_t int_32x4 = vmull_s16(int_16x4_x, int_16x4_in); - int_32x4 = vshlq_s32(int_32x4, int_32x4_scale); - int_32x4_sum = vaddq_s32(int_32x4_sum, int_32x4); - } - - // Use vector store to avoid long stall from data trasferring - // from vector to general register. - vst1q_s32(vbuff, int_32x4_sum); - csum32 = vbuff[0] + vbuff[1]; - csum32 += vbuff[2]; - csum32 += vbuff[3]; - } -#else - csum32 = 0; - if(scaling == 0) { - for (n = 0; n < PITCH_CORR_LEN2; n++) { - csum32 += x[n] * inptr[n]; - } - } else { - for (n = 0; n < PITCH_CORR_LEN2; n++) { - csum32 += (x[n] * inptr[n]) >> scaling; - } - } -#endif - - logcorQ8--; - - lys=Log2Q8((uint32_t)ysum32); // Q8 - lys=WEBRTC_SPL_RSHIFT_W32(lys, 1); //sqrt(ysum); - - if (csum32>0) { - - lcs=Log2Q8((uint32_t) csum32); // 2log(csum) in Q8 - - if (lcs>(lys + oneQ8) ){ // csum/sqrt(ysum) > 2 - *logcorQ8 = lcs - lys; // log2(csum/sqrt(ysum)) - } else { - *logcorQ8 = oneQ8; // 1.00 - } - - } else { - *logcorQ8 = 0; - } - } -} +extern void WebRtcIsacfix_PCorr2Q32(const int16_t *in, int32_t *logcorQ8); @@ -311,12 +209,13 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */ /* compute correlation for first and second half of the frame */ - PCorr2Q32(buf_dec16, crrvecQ8_1); - PCorr2Q32(buf_dec16 + PITCH_CORR_STEP2, crrvecQ8_2); + WebRtcIsacfix_PCorr2Q32(buf_dec16, crrvecQ8_1); + WebRtcIsacfix_PCorr2Q32(buf_dec16 + PITCH_CORR_STEP2, crrvecQ8_2); /* bias towards pitch lag of previous frame */ - tmp32a = Log2Q8((uint32_t) old_lagQ8) - 2304; // log2(0.5*oldlag) in Q8 + tmp32a = WebRtcIsacfix_Log2Q8((uint32_t) old_lagQ8) - 2304; + // log2(0.5*oldlag) in Q8 tmp32b = WEBRTC_SPL_MUL_16_16_RSFT(oldgQ12,oldgQ12, 10); //Q12 & * 4.0; gain_bias16 = (int16_t) tmp32b; //Q12 if (gain_bias16 > 3276) gain_bias16 = 3276; // 0.8 in Q12 @@ -325,7 +224,7 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */ for (k = 0; k < PITCH_LAG_SPAN2; k++) { if (crrvecQ8_1[k]>0) { - tmp32b = Log2Q8((uint32_t) (k + (PITCH_MIN_LAG/2-2))); + tmp32b = WebRtcIsacfix_Log2Q8((uint32_t) (k + (PITCH_MIN_LAG/2-2))); tmp16a = (int16_t) (tmp32b - tmp32a); // Q8 & fabs(ratio)<4 tmp32c = WEBRTC_SPL_MUL_16_16_RSFT(tmp16a,tmp16a, 6); //Q10 tmp16b = (int16_t) tmp32c; // Q10 & <8 @@ -334,7 +233,8 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */ tmp16d = Exp2Q10((int16_t) -tmp16c); //Q10 tmp32c = WEBRTC_SPL_MUL_16_16_RSFT(gain_bias16,tmp16d,13); // Q10 & * 0.5 bias16 = (int16_t) (1024 + tmp32c); // Q10 - tmp32b = Log2Q8((uint32_t) bias16) - 2560; // Q10 in -> Q8 out with 10*2^8 offset + tmp32b = WebRtcIsacfix_Log2Q8((uint32_t)bias16) - 2560; + // Q10 in -> Q8 out with 10*2^8 offset crrvecQ8_1[k] += tmp32b ; // -10*2^8 offset } } @@ -407,7 +307,7 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */ xq[0] = WEBRTC_SPL_LSHIFT_W32(xq[0], 8); Intrp1DQ8(xq, fxq, yq, fyq); - tmp32a= Log2Q8((uint32_t) *yq) - 2048; // offset 8*2^8 + tmp32a= WebRtcIsacfix_Log2Q8((uint32_t) *yq) - 2048; // offset 8*2^8 /* Bias towards short lags */ /* log(pow(0.8, log(2.0 * *y )))/log(2.0) */ tmp32b= WEBRTC_SPL_MUL_16_16_RSFT((int16_t) tmp32a, -42, 8); @@ -437,10 +337,13 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */ tmp32b = (int32_t) (WEBRTC_SPL_LSHIFT_W32(tmp32a, 1)) - ratq; // Q8 tmp32c = WEBRTC_SPL_MUL_16_16_RSFT((int16_t) tmp32b, (int16_t) tmp32b, 8); // Q8 - tmp32b = (int32_t) tmp32c + (int32_t) WEBRTC_SPL_RSHIFT_W32(ratq, 1); // (k-r)^2 + 0.5 * r Q8 - tmp32c = Log2Q8((uint32_t) tmp32a) - 2048; // offset 8*2^8 , log2(0.5*k) Q8 - tmp32d = Log2Q8((uint32_t) tmp32b) - 2048; // offset 8*2^8 , log2(0.5*k) Q8 - tmp32e = tmp32c -tmp32d; + tmp32b = (int32_t)tmp32c + (int32_t)WEBRTC_SPL_RSHIFT_W32(ratq, 1); + // (k-r)^2 + 0.5 * r Q8 + tmp32c = WebRtcIsacfix_Log2Q8((uint32_t)tmp32a) - 2048; + // offset 8*2^8 , log2(0.5*k) Q8 + tmp32d = WebRtcIsacfix_Log2Q8((uint32_t)tmp32b) - 2048; + // offset 8*2^8 , log2(0.5*k) Q8 + tmp32e = tmp32c - tmp32d; cv2q[k] += WEBRTC_SPL_RSHIFT_W32(tmp32e, 1); @@ -481,7 +384,7 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */ /* Bias towards short lags */ /* log(pow(0.8, log(2.0f * *y )))/log(2.0f) */ - tmp32a= Log2Q8((uint32_t) *yq) - 2048; // offset 8*2^8 + tmp32a= WebRtcIsacfix_Log2Q8((uint32_t) *yq) - 2048; // offset 8*2^8 tmp32b= WEBRTC_SPL_MUL_16_16_RSFT((int16_t) tmp32a, -82, 8); tmp32c= tmp32b + 256; *fyq += tmp32c; diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h index 93c81c8e..da401e5f 100644 --- a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h +++ b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h @@ -58,4 +58,8 @@ void WebRtcIsacfix_DecimateAllpass32(const int16_t *in, int16_t N, /* number of input samples */ int16_t *out); /* array of size N/2 */ +int32_t WebRtcIsacfix_Log2Q8( uint32_t x ); + +void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8); + #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c new file mode 100644 index 00000000..82155d27 --- /dev/null +++ b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c @@ -0,0 +1,122 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" + +#ifdef WEBRTC_ARCH_ARM_NEON +#include <arm_neon.h> +#endif + +#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" +#include "webrtc/system_wrappers/interface/compile_assert_c.h" + +extern int32_t WebRtcIsacfix_Log2Q8(uint32_t x); + +void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8) { + int16_t scaling,n,k; + int32_t ysum32,csum32, lys, lcs; + int32_t oneQ8; + const int16_t* x; + const int16_t* inptr; + + oneQ8 = WEBRTC_SPL_LSHIFT_W32((int32_t)1, 8); // 1.00 in Q8 + + x = in + PITCH_MAX_LAG / 2 + 2; + scaling = WebRtcSpl_GetScalingSquare((int16_t*)in, + PITCH_CORR_LEN2, + PITCH_CORR_LEN2); + ysum32 = 1; + csum32 = 0; + x = in + PITCH_MAX_LAG / 2 + 2; + for (n = 0; n < PITCH_CORR_LEN2; n++) { + ysum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)in[n], + (int16_t)in[n], + scaling); // Q0 + csum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)x[n], + (int16_t)in[n], + scaling); // Q0 + } + logcorQ8 += PITCH_LAG_SPAN2 - 1; + lys = WebRtcIsacfix_Log2Q8((uint32_t)ysum32); // Q8 + lys = WEBRTC_SPL_RSHIFT_W32(lys, 1); //sqrt(ysum); + if (csum32 > 0) { + lcs = WebRtcIsacfix_Log2Q8((uint32_t)csum32); // 2log(csum) in Q8 + if (lcs > (lys + oneQ8)) { // csum/sqrt(ysum) > 2 in Q8 + *logcorQ8 = lcs - lys; // log2(csum/sqrt(ysum)) + } else { + *logcorQ8 = oneQ8; // 1.00 + } + } else { + *logcorQ8 = 0; + } + + + for (k = 1; k < PITCH_LAG_SPAN2; k++) { + inptr = &in[k]; + ysum32 -= WEBRTC_SPL_MUL_16_16_RSFT((int16_t)in[k - 1], + (int16_t)in[k - 1], + scaling); + ysum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)in[PITCH_CORR_LEN2 + k - 1], + (int16_t)in[PITCH_CORR_LEN2 + k - 1], + scaling); +#ifdef WEBRTC_ARCH_ARM_NEON + { + int32_t vbuff[4]; + int32x4_t int_32x4_sum = vmovq_n_s32(0); + // Can't shift a Neon register to right with a non-constant shift value. + int32x4_t int_32x4_scale = vdupq_n_s32(-scaling); + // Assert a codition used in loop unrolling at compile-time. + COMPILE_ASSERT(PITCH_CORR_LEN2 %4 == 0); + + for (n = 0; n < PITCH_CORR_LEN2; n += 4) { + int16x4_t int_16x4_x = vld1_s16(&x[n]); + int16x4_t int_16x4_in = vld1_s16(&inptr[n]); + int32x4_t int_32x4 = vmull_s16(int_16x4_x, int_16x4_in); + int_32x4 = vshlq_s32(int_32x4, int_32x4_scale); + int_32x4_sum = vaddq_s32(int_32x4_sum, int_32x4); + } + + // Use vector store to avoid long stall from data trasferring + // from vector to general register. + vst1q_s32(vbuff, int_32x4_sum); + csum32 = vbuff[0] + vbuff[1]; + csum32 += vbuff[2]; + csum32 += vbuff[3]; + } +#else + csum32 = 0; + if(scaling == 0) { + for (n = 0; n < PITCH_CORR_LEN2; n++) { + csum32 += x[n] * inptr[n]; + } + } else { + for (n = 0; n < PITCH_CORR_LEN2; n++) { + csum32 += (x[n] * inptr[n]) >> scaling; + } + } +#endif + + logcorQ8--; + + lys = WebRtcIsacfix_Log2Q8((uint32_t)ysum32); // Q8 + lys = WEBRTC_SPL_RSHIFT_W32(lys, 1); //sqrt(ysum); + + if (csum32 > 0) { + lcs = WebRtcIsacfix_Log2Q8((uint32_t)csum32); // 2log(csum) in Q8 + if (lcs > (lys + oneQ8)) { // csum/sqrt(ysum) > 2 + *logcorQ8 = lcs - lys; // log2(csum/sqrt(ysum)) + } else { + *logcorQ8 = oneQ8; // 1.00 + } + } else { + *logcorQ8 = 0; + } + } +} diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c new file mode 100644 index 00000000..fa426e98 --- /dev/null +++ b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c @@ -0,0 +1,196 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" +#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" +#include "webrtc/system_wrappers/interface/compile_assert_c.h" + +extern int32_t WebRtcIsacfix_Log2Q8(uint32_t x); + +void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8) { + int16_t scaling,n,k; + int32_t ysum32,csum32, lys, lcs; + int32_t oneQ8; + const int16_t* x; + const int16_t* inptr; + + oneQ8 = WEBRTC_SPL_LSHIFT_W32((int32_t)1, 8); // 1.00 in Q8 + x = in + PITCH_MAX_LAG / 2 + 2; + scaling = WebRtcSpl_GetScalingSquare((int16_t*)in, + PITCH_CORR_LEN2, + PITCH_CORR_LEN2); + ysum32 = 1; + csum32 = 0; + x = in + PITCH_MAX_LAG / 2 + 2; + { + const int16_t* tmp_x = x; + const int16_t* tmp_in = in; + int32_t tmp1, tmp2, tmp3, tmp4, tmp5, tmp6, tmp7, tmp8; + n = PITCH_CORR_LEN2; + COMPILE_ASSERT(PITCH_CORR_LEN2 % 4 == 0); + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "1: \n\t" + "lh %[tmp1], 0(%[tmp_in]) \n\t" + "lh %[tmp2], 2(%[tmp_in]) \n\t" + "lh %[tmp3], 4(%[tmp_in]) \n\t" + "lh %[tmp4], 6(%[tmp_in]) \n\t" + "lh %[tmp5], 0(%[tmp_x]) \n\t" + "lh %[tmp6], 2(%[tmp_x]) \n\t" + "lh %[tmp7], 4(%[tmp_x]) \n\t" + "lh %[tmp8], 6(%[tmp_x]) \n\t" + "mul %[tmp5], %[tmp1], %[tmp5] \n\t" + "mul %[tmp1], %[tmp1], %[tmp1] \n\t" + "mul %[tmp6], %[tmp2], %[tmp6] \n\t" + "mul %[tmp2], %[tmp2], %[tmp2] \n\t" + "mul %[tmp7], %[tmp3], %[tmp7] \n\t" + "mul %[tmp3], %[tmp3], %[tmp3] \n\t" + "mul %[tmp8], %[tmp4], %[tmp8] \n\t" + "mul %[tmp4], %[tmp4], %[tmp4] \n\t" + "addiu %[n], %[n], -4 \n\t" + "srav %[tmp5], %[tmp5], %[scaling] \n\t" + "srav %[tmp1], %[tmp1], %[scaling] \n\t" + "srav %[tmp6], %[tmp6], %[scaling] \n\t" + "srav %[tmp2], %[tmp2], %[scaling] \n\t" + "srav %[tmp7], %[tmp7], %[scaling] \n\t" + "srav %[tmp3], %[tmp3], %[scaling] \n\t" + "srav %[tmp8], %[tmp8], %[scaling] \n\t" + "srav %[tmp4], %[tmp4], %[scaling] \n\t" + "addu %[ysum32], %[ysum32], %[tmp1] \n\t" + "addu %[csum32], %[csum32], %[tmp5] \n\t" + "addu %[ysum32], %[ysum32], %[tmp2] \n\t" + "addu %[csum32], %[csum32], %[tmp6] \n\t" + "addu %[ysum32], %[ysum32], %[tmp3] \n\t" + "addu %[csum32], %[csum32], %[tmp7] \n\t" + "addu %[ysum32], %[ysum32], %[tmp4] \n\t" + "addu %[csum32], %[csum32], %[tmp8] \n\t" + "addiu %[tmp_in], %[tmp_in], 8 \n\t" + "bgtz %[n], 1b \n\t" + " addiu %[tmp_x], %[tmp_x], 8 \n\t" + ".set pop \n\t" + : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [tmp3] "=&r" (tmp3), + [tmp4] "=&r" (tmp4), [tmp5] "=&r" (tmp5), [tmp6] "=&r" (tmp6), + [tmp7] "=&r" (tmp7), [tmp8] "=&r" (tmp8), [tmp_in] "+r" (tmp_in), + [ysum32] "+r" (ysum32), [tmp_x] "+r" (tmp_x), [csum32] "+r" (csum32), + [n] "+r" (n) + : [scaling] "r" (scaling) + : "memory", "hi", "lo" + ); + } + logcorQ8 += PITCH_LAG_SPAN2 - 1; + lys = WebRtcIsacfix_Log2Q8((uint32_t)ysum32); // Q8 + lys = WEBRTC_SPL_RSHIFT_W32(lys, 1); //sqrt(ysum); + if (csum32 > 0) { + lcs = WebRtcIsacfix_Log2Q8((uint32_t)csum32); // 2log(csum) in Q8 + if (lcs > (lys + oneQ8)) { // csum/sqrt(ysum) > 2 in Q8 + *logcorQ8 = lcs - lys; // log2(csum/sqrt(ysum)) + } else { + *logcorQ8 = oneQ8; // 1.00 + } + } else { + *logcorQ8 = 0; + } + + for (k = 1; k < PITCH_LAG_SPAN2; k++) { + inptr = &in[k]; + const int16_t* tmp_in1 = &in[k - 1]; + const int16_t* tmp_in2 = &in[PITCH_CORR_LEN2 + k - 1]; + const int16_t* tmp_x = x; + int32_t tmp1, tmp2, tmp3, tmp4, tmp5, tmp6, tmp7, tmp8; + n = PITCH_CORR_LEN2; + csum32 = 0; + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "lh %[tmp1], 0(%[tmp_in1]) \n\t" + "lh %[tmp2], 0(%[tmp_in2]) \n\t" + "mul %[tmp1], %[tmp1], %[tmp1] \n\t" + "mul %[tmp2], %[tmp2], %[tmp2] \n\t" + "srav %[tmp1], %[tmp1], %[scaling] \n\t" + "srav %[tmp2], %[tmp2], %[scaling] \n\t" + "subu %[ysum32], %[ysum32], %[tmp1] \n\t" + "bnez %[scaling], 2f \n\t" + " addu %[ysum32], %[ysum32], %[tmp2] \n\t" + "1: \n\t" + "lh %[tmp1], 0(%[inptr]) \n\t" + "lh %[tmp2], 0(%[tmp_x]) \n\t" + "lh %[tmp3], 2(%[inptr]) \n\t" + "lh %[tmp4], 2(%[tmp_x]) \n\t" + "lh %[tmp5], 4(%[inptr]) \n\t" + "lh %[tmp6], 4(%[tmp_x]) \n\t" + "lh %[tmp7], 6(%[inptr]) \n\t" + "lh %[tmp8], 6(%[tmp_x]) \n\t" + "mul %[tmp1], %[tmp1], %[tmp2] \n\t" + "mul %[tmp2], %[tmp3], %[tmp4] \n\t" + "mul %[tmp3], %[tmp5], %[tmp6] \n\t" + "mul %[tmp4], %[tmp7], %[tmp8] \n\t" + "addiu %[n], %[n], -4 \n\t" + "addiu %[inptr], %[inptr], 8 \n\t" + "addiu %[tmp_x], %[tmp_x], 8 \n\t" + "addu %[csum32], %[csum32], %[tmp1] \n\t" + "addu %[csum32], %[csum32], %[tmp2] \n\t" + "addu %[csum32], %[csum32], %[tmp3] \n\t" + "bgtz %[n], 1b \n\t" + " addu %[csum32], %[csum32], %[tmp4] \n\t" + "b 3f \n\t" + " nop \n\t" + "2: \n\t" + "lh %[tmp1], 0(%[inptr]) \n\t" + "lh %[tmp2], 0(%[tmp_x]) \n\t" + "lh %[tmp3], 2(%[inptr]) \n\t" + "lh %[tmp4], 2(%[tmp_x]) \n\t" + "lh %[tmp5], 4(%[inptr]) \n\t" + "lh %[tmp6], 4(%[tmp_x]) \n\t" + "lh %[tmp7], 6(%[inptr]) \n\t" + "lh %[tmp8], 6(%[tmp_x]) \n\t" + "mul %[tmp1], %[tmp1], %[tmp2] \n\t" + "mul %[tmp2], %[tmp3], %[tmp4] \n\t" + "mul %[tmp3], %[tmp5], %[tmp6] \n\t" + "mul %[tmp4], %[tmp7], %[tmp8] \n\t" + "addiu %[n], %[n], -4 \n\t" + "addiu %[inptr], %[inptr], 8 \n\t" + "addiu %[tmp_x], %[tmp_x], 8 \n\t" + "srav %[tmp1], %[tmp1], %[scaling] \n\t" + "srav %[tmp2], %[tmp2], %[scaling] \n\t" + "srav %[tmp3], %[tmp3], %[scaling] \n\t" + "srav %[tmp4], %[tmp4], %[scaling] \n\t" + "addu %[csum32], %[csum32], %[tmp1] \n\t" + "addu %[csum32], %[csum32], %[tmp2] \n\t" + "addu %[csum32], %[csum32], %[tmp3] \n\t" + "bgtz %[n], 2b \n\t" + " addu %[csum32], %[csum32], %[tmp4] \n\t" + "3: \n\t" + ".set pop \n\t" + : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [tmp3] "=&r" (tmp3), + [tmp4] "=&r" (tmp4), [tmp5] "=&r" (tmp5), [tmp6] "=&r" (tmp6), + [tmp7] "=&r" (tmp7), [tmp8] "=&r" (tmp8), [inptr] "+r" (inptr), + [csum32] "+r" (csum32), [tmp_x] "+r" (tmp_x), [ysum32] "+r" (ysum32), + [n] "+r" (n) + : [tmp_in1] "r" (tmp_in1), [tmp_in2] "r" (tmp_in2), + [scaling] "r" (scaling) + : "memory", "hi", "lo" + ); + + logcorQ8--; + lys = WebRtcIsacfix_Log2Q8((uint32_t)ysum32); // Q8 + lys = WEBRTC_SPL_RSHIFT_W32(lys, 1); //sqrt(ysum); + if (csum32 > 0) { + lcs = WebRtcIsacfix_Log2Q8((uint32_t)csum32); // 2log(csum) in Q8 + if (lcs > (lys + oneQ8)) { // csum/sqrt(ysum) > 2 + *logcorQ8 = lcs - lys; // log2(csum/sqrt(ysum)) + } else { + *logcorQ8 = oneQ8; // 1.00 + } + } else { + *logcorQ8 = 0; + } + } +} diff --git a/modules/audio_coding/codecs/isac/fix/source/transform.c b/modules/audio_coding/codecs/isac/fix/source/transform.c index 67e513c7..24ccc821 100644 --- a/modules/audio_coding/codecs/isac/fix/source/transform.c +++ b/modules/audio_coding/codecs/isac/fix/source/transform.c @@ -19,89 +19,13 @@ #include "webrtc/modules/audio_coding/codecs/isac/fix/source/fft.h" #include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h" -#if (defined WEBRTC_DETECT_ARM_NEON || defined WEBRTC_ARCH_ARM_NEON) -/* Tables are defined in ARM assembly files. */ +/* Tables are defined in transform_tables.c file or ARM assembly files. */ /* Cosine table 1 in Q14 */ extern const int16_t WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2]; /* Sine table 1 in Q14 */ extern const int16_t WebRtcIsacfix_kSinTab1[FRAMESAMPLES/2]; /* Sine table 2 in Q14 */ extern const int16_t WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4]; -#else -/* Cosine table 1 in Q14 */ -static const int16_t WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2] = { - 16384, 16383, 16378, 16371, 16362, 16349, 16333, 16315, 16294, 16270, - 16244, 16214, 16182, 16147, 16110, 16069, 16026, 15980, 15931, 15880, - 15826, 15769, 15709, 15647, 15582, 15515, 15444, 15371, 15296, 15218, - 15137, 15053, 14968, 14879, 14788, 14694, 14598, 14500, 14399, 14295, - 14189, 14081, 13970, 13856, 13741, 13623, 13502, 13380, 13255, 13128, - 12998, 12867, 12733, 12597, 12458, 12318, 12176, 12031, 11885, 11736, - 11585, 11433, 11278, 11121, 10963, 10803, 10641, 10477, 10311, 10143, - 9974, 9803, 9630, 9456, 9280, 9102, 8923, 8743, 8561, 8377, - 8192, 8006, 7818, 7629, 7438, 7246, 7053, 6859, 6664, 6467, - 6270, 6071, 5872, 5671, 5469, 5266, 5063, 4859, 4653, 4447, - 4240, 4033, 3825, 3616, 3406, 3196, 2986, 2775, 2563, 2351, - 2139, 1926, 1713, 1499, 1285, 1072, 857, 643, 429, 214, - 0, -214, -429, -643, -857, -1072, -1285, -1499, -1713, -1926, - -2139, -2351, -2563, -2775, -2986, -3196, -3406, -3616, -3825, -4033, - -4240, -4447, -4653, -4859, -5063, -5266, -5469, -5671, -5872, -6071, - -6270, -6467, -6664, -6859, -7053, -7246, -7438, -7629, -7818, -8006, - -8192, -8377, -8561, -8743, -8923, -9102, -9280, -9456, -9630, -9803, - -9974, -10143, -10311, -10477, -10641, -10803, -10963, -11121, -11278, -11433, - -11585, -11736, -11885, -12031, -12176, -12318, -12458, -12597, -12733, - -12867, -12998, -13128, -13255, -13380, -13502, -13623, -13741, -13856, - -13970, -14081, -14189, -14295, -14399, -14500, -14598, -14694, -14788, - -14879, -14968, -15053, -15137, -15218, -15296, -15371, -15444, -15515, - -15582, -15647, -15709, -15769, -15826, -15880, -15931, -15980, -16026, - -16069, -16110, -16147, -16182, -16214, -16244, -16270, -16294, -16315, - -16333, -16349, -16362, -16371, -16378, -16383 -}; - -/* Sine table 1 in Q14 */ -static const int16_t WebRtcIsacfix_kSinTab1[FRAMESAMPLES/2] = { - 0, 214, 429, 643, 857, 1072, 1285, 1499, 1713, 1926, - 2139, 2351, 2563, 2775, 2986, 3196, 3406, 3616, 3825, 4033, - 4240, 4447, 4653, 4859, 5063, 5266, 5469, 5671, 5872, 6071, - 6270, 6467, 6664, 6859, 7053, 7246, 7438, 7629, 7818, 8006, - 8192, 8377, 8561, 8743, 8923, 9102, 9280, 9456, 9630, 9803, - 9974, 10143, 10311, 10477, 10641, 10803, 10963, 11121, 11278, 11433, - 11585, 11736, 11885, 12031, 12176, 12318, 12458, 12597, 12733, 12867, - 12998, 13128, 13255, 13380, 13502, 13623, 13741, 13856, 13970, 14081, - 14189, 14295, 14399, 14500, 14598, 14694, 14788, 14879, 14968, 15053, - 15137, 15218, 15296, 15371, 15444, 15515, 15582, 15647, 15709, 15769, - 15826, 15880, 15931, 15980, 16026, 16069, 16110, 16147, 16182, 16214, - 16244, 16270, 16294, 16315, 16333, 16349, 16362, 16371, 16378, 16383, - 16384, 16383, 16378, 16371, 16362, 16349, 16333, 16315, 16294, 16270, - 16244, 16214, 16182, 16147, 16110, 16069, 16026, 15980, 15931, 15880, - 15826, 15769, 15709, 15647, 15582, 15515, 15444, 15371, 15296, 15218, - 15137, 15053, 14968, 14879, 14788, 14694, 14598, 14500, 14399, 14295, - 14189, 14081, 13970, 13856, 13741, 13623, 13502, 13380, 13255, 13128, - 12998, 12867, 12733, 12597, 12458, 12318, 12176, 12031, 11885, 11736, - 11585, 11433, 11278, 11121, 10963, 10803, 10641, 10477, 10311, 10143, - 9974, 9803, 9630, 9456, 9280, 9102, 8923, 8743, 8561, 8377, - 8192, 8006, 7818, 7629, 7438, 7246, 7053, 6859, 6664, 6467, - 6270, 6071, 5872, 5671, 5469, 5266, 5063, 4859, 4653, 4447, - 4240, 4033, 3825, 3616, 3406, 3196, 2986, 2775, 2563, 2351, - 2139, 1926, 1713, 1499, 1285, 1072, 857, 643, 429, 214 -}; - - -/* Sine table 2 in Q14 */ -static const int16_t WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4] = { - 16384, -16381, 16375, -16367, 16356, -16342, 16325, -16305, 16283, -16257, - 16229, -16199, 16165, -16129, 16090, -16048, 16003, -15956, 15906, -15853, - 15798, -15739, 15679, -15615, 15549, -15480, 15408, -15334, 15257, -15178, - 15095, -15011, 14924, -14834, 14741, -14647, 14549, -14449, 14347, -14242, - 14135, -14025, 13913, -13799, 13682, -13563, 13441, -13318, 13192, -13063, - 12933, -12800, 12665, -12528, 12389, -12247, 12104, -11958, 11810, -11661, - 11509, -11356, 11200, -11042, 10883, -10722, 10559, -10394, 10227, -10059, - 9889, -9717, 9543, -9368, 9191, -9013, 8833, -8652, 8469, -8285, - 8099, -7912, 7723, -7534, 7342, -7150, 6957, -6762, 6566, -6369, - 6171, -5971, 5771, -5570, 5368, -5165, 4961, -4756, 4550, -4344, - 4137, -3929, 3720, -3511, 3301, -3091, 2880, -2669, 2457, -2245, - 2032, -1819, 1606, -1392, 1179, -965, 750, -536, 322, -107 -}; -#endif // WEBRTC_DETECT_ARM_NEON || WEBRTC_ARCH_ARM_NEON void WebRtcIsacfix_Time2SpecC(int16_t *inre1Q9, int16_t *inre2Q9, diff --git a/modules/audio_coding/codecs/isac/fix/source/transform_mips.c b/modules/audio_coding/codecs/isac/fix/source/transform_mips.c new file mode 100644 index 00000000..bf95ee57 --- /dev/null +++ b/modules/audio_coding/codecs/isac/fix/source/transform_mips.c @@ -0,0 +1,1287 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h" +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/fft.h" +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h" + +// The tables are defined in transform_tables.c file. +extern const int16_t WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2]; +extern const int16_t WebRtcIsacfix_kSinTab1[FRAMESAMPLES/2]; +extern const int16_t WebRtcIsacfix_kCosTab2[FRAMESAMPLES/4]; +extern const int16_t WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4]; + +// MIPS DSPr2 version of the WebRtcIsacfix_Time2Spec function +// is not bit-exact with the C version. +// The accuracy of the MIPS DSPr2 version is same or better. +void WebRtcIsacfix_Time2SpecMIPS(int16_t* inre1Q9, + int16_t* inre2Q9, + int16_t* outreQ7, + int16_t* outimQ7) { + int k = FRAMESAMPLES / 2; + int32_t tmpreQ16[FRAMESAMPLES / 2], tmpimQ16[FRAMESAMPLES / 2]; + int32_t r0, r1, r2, r3, r4, r5, r6, r7, r8, r9; + int32_t inre1, inre2, tmpre, tmpim, factor, max, max1; + int16_t* cosptr; + int16_t* sinptr; + + cosptr = (int16_t*)WebRtcIsacfix_kCosTab1; + sinptr = (int16_t*)WebRtcIsacfix_kSinTab1; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[inre1], %[inre1Q9], 0 \n\t" + "addiu %[inre2], %[inre2Q9], 0 \n\t" + "addiu %[tmpre], %[tmpreQ16], 0 \n\t" + "addiu %[tmpim], %[tmpimQ16], 0 \n\t" + "addiu %[factor], $zero, 16921 \n\t" + "mul %[max], $zero, $zero \n\t" + // Multiply with complex exponentials and combine into one complex vector. + // Also, calculate the maximal absolute value in the same loop. + "1: \n\t" +#if defined(MIPS_DSP_R2_LE) + "lwl %[r0], 0(%[inre1]) \n\t" + "lwl %[r2], 0(%[cosptr]) \n\t" + "lwl %[r3], 0(%[sinptr]) \n\t" + "lwl %[r1], 0(%[inre2]) \n\t" + "lwr %[r0], 0(%[inre1]) \n\t" + "lwr %[r2], 0(%[cosptr]) \n\t" + "lwr %[r3], 0(%[sinptr]) \n\t" + "lwr %[r1], 0(%[inre2]) \n\t" + "muleq_s.w.phr %[r4], %[r2], %[r0] \n\t" + "muleq_s.w.phr %[r5], %[r3], %[r0] \n\t" + "muleq_s.w.phr %[r6], %[r3], %[r1] \n\t" + "muleq_s.w.phr %[r7], %[r2], %[r1] \n\t" + "muleq_s.w.phl %[r8], %[r2], %[r0] \n\t" + "muleq_s.w.phl %[r0], %[r3], %[r0] \n\t" + "muleq_s.w.phl %[r3], %[r3], %[r1] \n\t" + "muleq_s.w.phl %[r1], %[r2], %[r1] \n\t" + "addiu %[k], %[k], -2 \n\t" + "addu %[r4], %[r4], %[r6] \n\t" + "subu %[r5], %[r7], %[r5] \n\t" + "sra %[r4], %[r4], 8 \n\t" + "sra %[r5], %[r5], 8 \n\t" + "mult $ac0, %[factor], %[r4] \n\t" + "mult $ac1, %[factor], %[r5] \n\t" + "addu %[r3], %[r8], %[r3] \n\t" + "subu %[r0], %[r1], %[r0] \n\t" + "sra %[r3], %[r3], 8 \n\t" + "sra %[r0], %[r0], 8 \n\t" + "mult $ac2, %[factor], %[r3] \n\t" + "mult $ac3, %[factor], %[r0] \n\t" + "extr_r.w %[r4], $ac0, 16 \n\t" + "extr_r.w %[r5], $ac1, 16 \n\t" + "addiu %[inre1], %[inre1], 4 \n\t" + "addiu %[inre2], %[inre2], 4 \n\t" + "extr_r.w %[r6], $ac2, 16 \n\t" + "extr_r.w %[r7], $ac3, 16 \n\t" + "addiu %[cosptr], %[cosptr], 4 \n\t" + "addiu %[sinptr], %[sinptr], 4 \n\t" + "shra_r.w %[r4], %[r4], 3 \n\t" + "shra_r.w %[r5], %[r5], 3 \n\t" + "sw %[r4], 0(%[tmpre]) \n\t" + "absq_s.w %[r4], %[r4] \n\t" + "sw %[r5], 0(%[tmpim]) \n\t" + "absq_s.w %[r5], %[r5] \n\t" + "shra_r.w %[r6], %[r6], 3 \n\t" + "shra_r.w %[r7], %[r7], 3 \n\t" + "sw %[r6], 4(%[tmpre]) \n\t" + "absq_s.w %[r6], %[r6] \n\t" + "sw %[r7], 4(%[tmpim]) \n\t" + "absq_s.w %[r7], %[r7] \n\t" + "slt %[r0], %[r4], %[r5] \n\t" + "movn %[r4], %[r5], %[r0] \n\t" + "slt %[r1], %[r6], %[r7] \n\t" + "movn %[r6], %[r7], %[r1] \n\t" + "slt %[r0], %[max], %[r4] \n\t" + "movn %[max], %[r4], %[r0] \n\t" + "slt %[r1], %[max], %[r6] \n\t" + "movn %[max], %[r6], %[r1] \n\t" + "addiu %[tmpre], %[tmpre], 8 \n\t" + "bgtz %[k], 1b \n\t" + " addiu %[tmpim], %[tmpim], 8 \n\t" +#else // #if defined(MIPS_DSP_R2_LE) + "lh %[r0], 0(%[inre1]) \n\t" + "lh %[r1], 0(%[inre2]) \n\t" + "lh %[r2], 0(%[cosptr]) \n\t" + "lh %[r3], 0(%[sinptr]) \n\t" + "addiu %[k], %[k], -1 \n\t" + "mul %[r4], %[r0], %[r2] \n\t" + "mul %[r5], %[r1], %[r3] \n\t" + "mul %[r0], %[r0], %[r3] \n\t" + "mul %[r2], %[r1], %[r2] \n\t" + "addiu %[inre1], %[inre1], 2 \n\t" + "addiu %[inre2], %[inre2], 2 \n\t" + "addiu %[cosptr], %[cosptr], 2 \n\t" + "addiu %[sinptr], %[sinptr], 2 \n\t" + "addu %[r1], %[r4], %[r5] \n\t" + "sra %[r1], %[r1], 7 \n\t" + "sra %[r3], %[r1], 16 \n\t" + "andi %[r1], %[r1], 0xFFFF \n\t" + "sra %[r1], %[r1], 1 \n\t" + "mul %[r1], %[factor], %[r1] \n\t" + "mul %[r3], %[factor], %[r3] \n\t" + "subu %[r0], %[r2], %[r0] \n\t" + "sra %[r0], %[r0], 7 \n\t" + "sra %[r2], %[r0], 16 \n\t" + "andi %[r0], %[r0], 0xFFFF \n\t" + "sra %[r0], %[r0], 1 \n\t" + "mul %[r0], %[factor], %[r0] \n\t" + "mul %[r2], %[factor], %[r2] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r1], %[r1], 15 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r1], %[r1], 0x4000 \n\t" + "sra %[r1], %[r1], 15 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "addu %[r1], %[r3], %[r1] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r1], %[r1], 3 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r1], %[r1], 4 \n\t" + "sra %[r1], %[r1], 3 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sw %[r1], 0(%[tmpre]) \n\t" + "addiu %[tmpre], %[tmpre], 4 \n\t" +#if defined(MIPS_DSP_R1_LE) + "absq_s.w %[r1], %[r1] \n\t" + "shra_r.w %[r0], %[r0], 15 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "negu %[r4], %[r1] \n\t" + "slt %[r3], %[r1], $zero \n\t" + "movn %[r1], %[r4], %[r3] \n\t" + "addiu %[r0], %[r0], 0x4000 \n\t" + "sra %[r0], %[r0], 15 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "addu %[r0], %[r0], %[r2] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r0], %[r0], 3 \n\t" + "sw %[r0], 0(%[tmpim]) \n\t" + "absq_s.w %[r0], %[r0] \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r0], %[r0], 4 \n\t" + "sra %[r0], %[r0], 3 \n\t" + "sw %[r0], 0(%[tmpim]) \n\t" + "negu %[r2], %[r0] \n\t" + "slt %[r3], %[r0], $zero \n\t" + "movn %[r0], %[r2], %[r3] \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "slt %[r2], %[max], %[r1] \n\t" + "movn %[max], %[r1], %[r2] \n\t" + "slt %[r2], %[max], %[r0] \n\t" + "movn %[max], %[r0], %[r2] \n\t" + "bgtz %[k], 1b \n\t" + " addiu %[tmpim], %[tmpim], 4 \n\t" +#endif // #if defined(MIPS_DSP_R2_LE) + // Calculate WebRtcSpl_NormW32(max). + // If max gets value >=0, we should shift max steps to the left, and the + // domain will be Q(16+shift). If max gets value <0, we should shift -max + // steps to the right, and the domain will be Q(16+max) + "clz %[max], %[max] \n\t" + "addiu %[max], %[max], -25 \n\t" + ".set pop \n\t" + : [k] "+r" (k), [inre1] "=&r" (inre1), [inre2] "=&r" (inre2), + [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), + [r3] "=&r" (r3), [r4] "=&r" (r4), [tmpre] "=&r" (tmpre), + [tmpim] "=&r" (tmpim), [max] "=&r" (max), [factor] "=&r" (factor), +#if defined(MIPS_DSP_R2_LE) + [r6] "=&r" (r6), [r7] "=&r" (r7), [r8] "=&r" (r8), +#endif // #if defined(MIPS_DSP_R2_LE) + [r5] "=&r" (r5) + : [inre1Q9] "r" (inre1Q9), [inre2Q9] "r" (inre2Q9), + [tmpreQ16] "r" (tmpreQ16), [tmpimQ16] "r" (tmpimQ16), + [cosptr] "r" (cosptr), [sinptr] "r" (sinptr) + : "hi", "lo", "memory" + ); + + // "Fastest" vectors + k = FRAMESAMPLES / 4; + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[tmpre], %[tmpreQ16], 0 \n\t" + "addiu %[tmpim], %[tmpimQ16], 0 \n\t" + "addiu %[inre1], %[inre1Q9], 0 \n\t" + "addiu %[inre2], %[inre2Q9], 0 \n\t" + "blez %[max], 2f \n\t" + " subu %[max1], $zero, %[max] \n\t" + "1: \n\t" + "lw %[r0], 0(%[tmpre]) \n\t" + "lw %[r1], 0(%[tmpim]) \n\t" + "lw %[r2], 4(%[tmpre]) \n\t" + "lw %[r3], 4(%[tmpim]) \n\t" + "addiu %[k], %[k], -1 \n\t" + "sllv %[r0], %[r0], %[max] \n\t" + "sllv %[r1], %[r1], %[max] \n\t" + "sllv %[r2], %[r2], %[max] \n\t" + "sllv %[r3], %[r3], %[max] \n\t" + "addiu %[tmpre], %[tmpre], 8 \n\t" + "addiu %[tmpim], %[tmpim], 8 \n\t" + "sh %[r0], 0(%[inre1]) \n\t" + "sh %[r1], 0(%[inre2]) \n\t" + "sh %[r2], 2(%[inre1]) \n\t" + "sh %[r3], 2(%[inre2]) \n\t" + "addiu %[inre1], %[inre1], 4 \n\t" + "bgtz %[k], 1b \n\t" + " addiu %[inre2], %[inre2], 4 \n\t" + "b 4f \n\t" + " nop \n\t" + "2: \n\t" +#if !defined(MIPS_DSP_R1_LE) + "addiu %[r4], %[max1], -1 \n\t" + "addiu %[r5], $zero, 1 \n\t" + "sllv %[r4], %[r5], %[r4] \n\t" +#endif // #if !defined(MIPS_DSP_R1_LE) + "3: \n\t" + "lw %[r0], 0(%[tmpre]) \n\t" + "lw %[r1], 0(%[tmpim]) \n\t" + "lw %[r2], 4(%[tmpre]) \n\t" + "lw %[r3], 4(%[tmpim]) \n\t" + "addiu %[k], %[k], -1 \n\t" +#if defined(MIPS_DSP_R1_LE) + "shrav_r.w %[r0], %[r0], %[max1] \n\t" + "shrav_r.w %[r1], %[r1], %[max1] \n\t" + "shrav_r.w %[r2], %[r2], %[max1] \n\t" + "shrav_r.w %[r3], %[r3], %[max1] \n\t" +#else // #if !defined(MIPS_DSP_R1_LE) + "addu %[r0], %[r0], %[r4] \n\t" + "addu %[r1], %[r1], %[r4] \n\t" + "addu %[r2], %[r2], %[r4] \n\t" + "addu %[r3], %[r3], %[r4] \n\t" + "srav %[r0], %[r0], %[max1] \n\t" + "srav %[r1], %[r1], %[max1] \n\t" + "srav %[r2], %[r2], %[max1] \n\t" + "srav %[r3], %[r3], %[max1] \n\t" +#endif // #if !defined(MIPS_DSP_R1_LE) + "addiu %[tmpre], %[tmpre], 8 \n\t" + "addiu %[tmpim], %[tmpim], 8 \n\t" + "sh %[r0], 0(%[inre1]) \n\t" + "sh %[r1], 0(%[inre2]) \n\t" + "sh %[r2], 2(%[inre1]) \n\t" + "sh %[r3], 2(%[inre2]) \n\t" + "addiu %[inre1], %[inre1], 4 \n\t" + "bgtz %[k], 3b \n\t" + " addiu %[inre2], %[inre2], 4 \n\t" + "4: \n\t" + ".set pop \n\t" + : [tmpre] "=&r" (tmpre), [tmpim] "=&r" (tmpim), [inre1] "=&r" (inre1), + [inre2] "=&r" (inre2), [k] "+r" (k), [max1] "=&r" (max1), +#if !defined(MIPS_DSP_R1_LE) + [r4] "=&r" (r4), [r5] "=&r" (r5), +#endif // #if !defined(MIPS_DSP_R1_LE) + [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3) + : [tmpreQ16] "r" (tmpreQ16), [tmpimQ16] "r" (tmpimQ16), + [inre1Q9] "r" (inre1Q9), [inre2Q9] "r" (inre2Q9), [max] "r" (max) + : "memory" + ); + + // Get DFT + WebRtcIsacfix_FftRadix16Fastest(inre1Q9, inre2Q9, -1); // real call + + // "Fastest" vectors and + // Use symmetry to separate into two complex vectors + // and center frames in time around zero + // merged into one loop + cosptr = (int16_t*)WebRtcIsacfix_kCosTab2; + sinptr = (int16_t*)WebRtcIsacfix_kSinTab2; + k = FRAMESAMPLES / 4; + factor = FRAMESAMPLES - 2; // offset for FRAMESAMPLES / 2 - 1 array member + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[inre1], %[inre1Q9], 0 \n\t" + "addiu %[inre2], %[inre2Q9], 0 \n\t" + "addiu %[tmpre], %[outreQ7], 0 \n\t" + "addiu %[tmpim], %[outimQ7], 0 \n\t" + "bltz %[max], 2f \n\t" + " subu %[max1], $zero, %[max] \n\t" + "1: \n\t" +#if !defined(MIPS_DSP_R1_LE) + "addu %[r4], %[inre1], %[offset] \n\t" + "addu %[r5], %[inre2], %[offset] \n\t" +#endif // #if !defined(MIPS_DSP_R1_LE) + "lh %[r0], 0(%[inre1]) \n\t" + "lh %[r1], 0(%[inre2]) \n\t" +#if defined(MIPS_DSP_R1_LE) + "lhx %[r2], %[offset](%[inre1]) \n\t" + "lhx %[r3], %[offset](%[inre2]) \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "lh %[r2], 0(%[r4]) \n\t" + "lh %[r3], 0(%[r5]) \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "srav %[r0], %[r0], %[max] \n\t" + "srav %[r1], %[r1], %[max] \n\t" + "srav %[r2], %[r2], %[max] \n\t" + "srav %[r3], %[r3], %[max] \n\t" + "addu %[r4], %[r0], %[r2] \n\t" + "subu %[r0], %[r2], %[r0] \n\t" + "subu %[r2], %[r1], %[r3] \n\t" + "addu %[r1], %[r1], %[r3] \n\t" + "lh %[r3], 0(%[cosptr]) \n\t" + "lh %[r5], 0(%[sinptr]) \n\t" + "andi %[r6], %[r4], 0xFFFF \n\t" + "sra %[r4], %[r4], 16 \n\t" + "mul %[r7], %[r3], %[r6] \n\t" + "mul %[r8], %[r3], %[r4] \n\t" + "mul %[r6], %[r5], %[r6] \n\t" + "mul %[r4], %[r5], %[r4] \n\t" + "addiu %[k], %[k], -1 \n\t" + "addiu %[inre1], %[inre1], 2 \n\t" + "addiu %[inre2], %[inre2], 2 \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r7], %[r7], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r7], %[r7], 0x2000 \n\t" + "sra %[r7], %[r7], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r8], %[r8], 2 \n\t" + "addu %[r8], %[r8], %[r7] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r6], %[r6], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r6], %[r6], 0x2000 \n\t" + "sra %[r6], %[r6], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r4], %[r4], 2 \n\t" + "addu %[r4], %[r4], %[r6] \n\t" + "andi %[r6], %[r2], 0xFFFF \n\t" + "sra %[r2], %[r2], 16 \n\t" + "mul %[r7], %[r5], %[r6] \n\t" + "mul %[r9], %[r5], %[r2] \n\t" + "mul %[r6], %[r3], %[r6] \n\t" + "mul %[r2], %[r3], %[r2] \n\t" + "addiu %[cosptr], %[cosptr], 2 \n\t" + "addiu %[sinptr], %[sinptr], 2 \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r7], %[r7], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r7], %[r7], 0x2000 \n\t" + "sra %[r7], %[r7], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r9], %[r9], 2 \n\t" + "addu %[r9], %[r7], %[r9] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r6], %[r6], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r6], %[r6], 0x2000 \n\t" + "sra %[r6], %[r6], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r2], %[r2], 2 \n\t" + "addu %[r2], %[r6], %[r2] \n\t" + "subu %[r8], %[r8], %[r9] \n\t" + "sra %[r8], %[r8], 9 \n\t" + "addu %[r2], %[r4], %[r2] \n\t" + "sra %[r2], %[r2], 9 \n\t" + "sh %[r8], 0(%[tmpre]) \n\t" + "sh %[r2], 0(%[tmpim]) \n\t" + + "andi %[r4], %[r1], 0xFFFF \n\t" + "sra %[r1], %[r1], 16 \n\t" + "andi %[r6], %[r0], 0xFFFF \n\t" + "sra %[r0], %[r0], 16 \n\t" + "mul %[r7], %[r5], %[r4] \n\t" + "mul %[r9], %[r5], %[r1] \n\t" + "mul %[r4], %[r3], %[r4] \n\t" + "mul %[r1], %[r3], %[r1] \n\t" + "mul %[r8], %[r3], %[r0] \n\t" + "mul %[r3], %[r3], %[r6] \n\t" + "mul %[r6], %[r5], %[r6] \n\t" + "mul %[r0], %[r5], %[r0] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r7], %[r7], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r7], %[r7], 0x2000 \n\t" + "sra %[r7], %[r7], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r9], %[r9], 2 \n\t" + "addu %[r9], %[r9], %[r7] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r4], %[r4], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r4], %[r4], 0x2000 \n\t" + "sra %[r4], %[r4], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r1], %[r1], 2 \n\t" + "addu %[r1], %[r1], %[r4] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r3], %[r3], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r3], %[r3], 0x2000 \n\t" + "sra %[r3], %[r3], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r8], %[r8], 2 \n\t" + "addu %[r8], %[r8], %[r3] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r6], %[r6], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r6], %[r6], 0x2000 \n\t" + "sra %[r6], %[r6], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r0], %[r0], 2 \n\t" + "addu %[r0], %[r0], %[r6] \n\t" + "addu %[r3], %[tmpre], %[offset] \n\t" + "addu %[r2], %[tmpim], %[offset] \n\t" + "addu %[r9], %[r9], %[r8] \n\t" + "negu %[r9], %[r9] \n\t" + "sra %[r9], %[r9], 9 \n\t" + "subu %[r0], %[r0], %[r1] \n\t" + "addiu %[offset], %[offset], -4 \n\t" + "sh %[r9], 0(%[r3]) \n\t" + "sh %[r0], 0(%[r2]) \n\t" + "addiu %[tmpre], %[tmpre], 2 \n\t" + "bgtz %[k], 1b \n\t" + " addiu %[tmpim], %[tmpim], 2 \n\t" + "b 3f \n\t" + " nop \n\t" + "2: \n\t" +#if !defined(MIPS_DSP_R1_LE) + "addu %[r4], %[inre1], %[offset] \n\t" + "addu %[r5], %[inre2], %[offset] \n\t" +#endif // #if !defined(MIPS_DSP_R1_LE) + "lh %[r0], 0(%[inre1]) \n\t" + "lh %[r1], 0(%[inre2]) \n\t" +#if defined(MIPS_DSP_R1_LE) + "lhx %[r2], %[offset](%[inre1]) \n\t" + "lhx %[r3], %[offset](%[inre2]) \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "lh %[r2], 0(%[r4]) \n\t" + "lh %[r3], 0(%[r5]) \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sllv %[r0], %[r0], %[max1] \n\t" + "sllv %[r1], %[r1], %[max1] \n\t" + "sllv %[r2], %[r2], %[max1] \n\t" + "sllv %[r3], %[r3], %[max1] \n\t" + "addu %[r4], %[r0], %[r2] \n\t" + "subu %[r0], %[r2], %[r0] \n\t" + "subu %[r2], %[r1], %[r3] \n\t" + "addu %[r1], %[r1], %[r3] \n\t" + "lh %[r3], 0(%[cosptr]) \n\t" + "lh %[r5], 0(%[sinptr]) \n\t" + "andi %[r6], %[r4], 0xFFFF \n\t" + "sra %[r4], %[r4], 16 \n\t" + "mul %[r7], %[r3], %[r6] \n\t" + "mul %[r8], %[r3], %[r4] \n\t" + "mul %[r6], %[r5], %[r6] \n\t" + "mul %[r4], %[r5], %[r4] \n\t" + "addiu %[k], %[k], -1 \n\t" + "addiu %[inre1], %[inre1], 2 \n\t" + "addiu %[inre2], %[inre2], 2 \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r7], %[r7], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r7], %[r7], 0x2000 \n\t" + "sra %[r7], %[r7], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r8], %[r8], 2 \n\t" + "addu %[r8], %[r8], %[r7] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r6], %[r6], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r6], %[r6], 0x2000 \n\t" + "sra %[r6], %[r6], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r4], %[r4], 2 \n\t" + "addu %[r4], %[r4], %[r6] \n\t" + "andi %[r6], %[r2], 0xFFFF \n\t" + "sra %[r2], %[r2], 16 \n\t" + "mul %[r7], %[r5], %[r6] \n\t" + "mul %[r9], %[r5], %[r2] \n\t" + "mul %[r6], %[r3], %[r6] \n\t" + "mul %[r2], %[r3], %[r2] \n\t" + "addiu %[cosptr], %[cosptr], 2 \n\t" + "addiu %[sinptr], %[sinptr], 2 \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r7], %[r7], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r7], %[r7], 0x2000 \n\t" + "sra %[r7], %[r7], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r9], %[r9], 2 \n\t" + "addu %[r9], %[r7], %[r9] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r6], %[r6], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r6], %[r6], 0x2000 \n\t" + "sra %[r6], %[r6], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r2], %[r2], 2 \n\t" + "addu %[r2], %[r6], %[r2] \n\t" + "subu %[r8], %[r8], %[r9] \n\t" + "sra %[r8], %[r8], 9 \n\t" + "addu %[r2], %[r4], %[r2] \n\t" + "sra %[r2], %[r2], 9 \n\t" + "sh %[r8], 0(%[tmpre]) \n\t" + "sh %[r2], 0(%[tmpim]) \n\t" + "andi %[r4], %[r1], 0xFFFF \n\t" + "sra %[r1], %[r1], 16 \n\t" + "andi %[r6], %[r0], 0xFFFF \n\t" + "sra %[r0], %[r0], 16 \n\t" + "mul %[r7], %[r5], %[r4] \n\t" + "mul %[r9], %[r5], %[r1] \n\t" + "mul %[r4], %[r3], %[r4] \n\t" + "mul %[r1], %[r3], %[r1] \n\t" + "mul %[r8], %[r3], %[r0] \n\t" + "mul %[r3], %[r3], %[r6] \n\t" + "mul %[r6], %[r5], %[r6] \n\t" + "mul %[r0], %[r5], %[r0] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r7], %[r7], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r7], %[r7], 0x2000 \n\t" + "sra %[r7], %[r7], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r9], %[r9], 2 \n\t" + "addu %[r9], %[r9], %[r7] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r4], %[r4], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r4], %[r4], 0x2000 \n\t" + "sra %[r4], %[r4], 14 \n\t" +#endif + "sll %[r1], %[r1], 2 \n\t" + "addu %[r1], %[r1], %[r4] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r3], %[r3], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r3], %[r3], 0x2000 \n\t" + "sra %[r3], %[r3], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r8], %[r8], 2 \n\t" + "addu %[r8], %[r8], %[r3] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r6], %[r6], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r6], %[r6], 0x2000 \n\t" + "sra %[r6], %[r6], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "sll %[r0], %[r0], 2 \n\t" + "addu %[r0], %[r0], %[r6] \n\t" + "addu %[r3], %[tmpre], %[offset] \n\t" + "addu %[r2], %[tmpim], %[offset] \n\t" + "addu %[r9], %[r9], %[r8] \n\t" + "negu %[r9], %[r9] \n\t" + "sra %[r9], %[r9], 9 \n\t" + "subu %[r0], %[r0], %[r1] \n\t" + "sra %[r0], %[r0], 9 \n\t" + "addiu %[offset], %[offset], -4 \n\t" + "sh %[r9], 0(%[r3]) \n\t" + "sh %[r0], 0(%[r2]) \n\t" + "addiu %[tmpre], %[tmpre], 2 \n\t" + "bgtz %[k], 2b \n\t" + " addiu %[tmpim], %[tmpim], 2 \n\t" + "3: \n\t" + ".set pop \n\t" + : [inre1] "=&r" (inre1), [inre2] "=&r" (inre2), [tmpre] "=&r" (tmpre), + [tmpim] "=&r" (tmpim), [offset] "+r" (factor), [k] "+r" (k), + [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), + [r4] "=&r" (r4), [r5] "=&r" (r5), [r6] "=&r" (r6), [r7] "=&r" (r7), + [r8] "=&r" (r8), [r9] "=&r" (r9), [max1] "=&r" (max1) + : [inre1Q9] "r" (inre1Q9), [inre2Q9] "r" (inre2Q9), + [outreQ7] "r" (outreQ7), [outimQ7] "r" (outimQ7), + [max] "r" (max), [cosptr] "r" (cosptr), [sinptr] "r" (sinptr) + : "hi", "lo", "memory" + ); +} + +void WebRtcIsacfix_Spec2TimeMIPS(int16_t *inreQ7, + int16_t *inimQ7, + int32_t *outre1Q16, + int32_t *outre2Q16) { + int k = FRAMESAMPLES / 4; + int16_t* inre; + int16_t* inim; + int32_t* outre1; + int32_t* outre2; + int16_t* cosptr = (int16_t*)WebRtcIsacfix_kCosTab2; + int16_t* sinptr = (int16_t*)WebRtcIsacfix_kSinTab2; + int32_t r0, r1, r2, r3, r4, r5, r6, r7, r8, r9, r10, max, max1; +#if defined(MIPS_DSP_R1_LE) + int32_t offset = FRAMESAMPLES - 4; +#else // #if defined(MIPS_DSP_R1_LE) + int32_t offset = FRAMESAMPLES - 2; +#endif // #if defined(MIPS_DSP_R1_LE) + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[inre], %[inreQ7], 0 \n\t" + "addiu %[inim] , %[inimQ7], 0 \n\t" + "addiu %[outre1], %[outre1Q16], 0 \n\t" + "addiu %[outre2], %[outre2Q16], 0 \n\t" + "mul %[max], $zero, $zero \n\t" + "1: \n\t" +#if defined(MIPS_DSP_R1_LE) + // Process two samples in one iteration avoiding left shift before + // multiplication. MaxAbsValueW32 function inlined into the loop. + "addu %[r8], %[inre], %[offset] \n\t" + "addu %[r9], %[inim], %[offset] \n\t" + "lwl %[r4], 0(%[r8]) \n\t" + "lwl %[r5], 0(%[r9]) \n\t" + "lwl %[r0], 0(%[inre]) \n\t" + "lwl %[r1], 0(%[inim]) \n\t" + "lwl %[r2], 0(%[cosptr]) \n\t" + "lwl %[r3], 0(%[sinptr]) \n\t" + "lwr %[r4], 0(%[r8]) \n\t" + "lwr %[r5], 0(%[r9]) \n\t" + "lwr %[r0], 0(%[inre]) \n\t" + "lwr %[r1], 0(%[inim]) \n\t" + "lwr %[r2], 0(%[cosptr]) \n\t" + "lwr %[r3], 0(%[sinptr]) \n\t" + "packrl.ph %[r4], %[r4], %[r4] \n\t" + "packrl.ph %[r5], %[r5], %[r5] \n\t" + "muleq_s.w.phr %[r6], %[r0], %[r2] \n\t" + "muleq_s.w.phr %[r7], %[r1], %[r3] \n\t" + "muleq_s.w.phr %[r8], %[r4], %[r2] \n\t" + "muleq_s.w.phr %[r9], %[r5], %[r3] \n\t" + "addiu %[k], %[k], -2 \n\t" + "addiu %[cosptr], %[cosptr], 4 \n\t" + "addiu %[sinptr], %[sinptr], 4 \n\t" + "addiu %[inre], %[inre], 4 \n\t" + "addiu %[inim], %[inim], 4 \n\t" + "shra_r.w %[r6], %[r6], 6 \n\t" + "shra_r.w %[r7], %[r7], 6 \n\t" + "shra_r.w %[r8], %[r8], 6 \n\t" + "shra_r.w %[r9], %[r9], 6 \n\t" + "addu %[r6], %[r6], %[r7] \n\t" + "subu %[r9], %[r9], %[r8] \n\t" + "subu %[r7], %[r6], %[r9] \n\t" + "addu %[r6], %[r6], %[r9] \n\t" + "sll %[r10], %[offset], 1 \n\t" + "addu %[r10], %[outre1], %[r10] \n\t" + "sw %[r7], 0(%[outre1]) \n\t" + "absq_s.w %[r7], %[r7] \n\t" + "sw %[r6], 4(%[r10]) \n\t" + "absq_s.w %[r6], %[r6] \n\t" + "slt %[r8], %[max], %[r7] \n\t" + "movn %[max], %[r7], %[r8] \n\t" + "slt %[r8], %[max], %[r6] \n\t" + "movn %[max], %[r6], %[r8] \n\t" + "muleq_s.w.phl %[r6], %[r0], %[r2] \n\t" + "muleq_s.w.phl %[r7], %[r1], %[r3] \n\t" + "muleq_s.w.phl %[r8], %[r4], %[r2] \n\t" + "muleq_s.w.phl %[r9], %[r5], %[r3] \n\t" + "shra_r.w %[r6], %[r6], 6 \n\t" + "shra_r.w %[r7], %[r7], 6 \n\t" + "shra_r.w %[r8], %[r8], 6 \n\t" + "shra_r.w %[r9], %[r9], 6 \n\t" + "addu %[r6], %[r6], %[r7] \n\t" + "subu %[r9], %[r9], %[r8] \n\t" + "subu %[r7], %[r6], %[r9] \n\t" + "addu %[r6], %[r6], %[r9] \n\t" + "sw %[r7], 4(%[outre1]) \n\t" + "absq_s.w %[r7], %[r7] \n\t" + "sw %[r6], 0(%[r10]) \n\t" + "absq_s.w %[r6], %[r6] \n\t" + "slt %[r8], %[max], %[r7] \n\t" + "movn %[max], %[r7], %[r8] \n\t" + "slt %[r8], %[max], %[r6] \n\t" + "movn %[max], %[r6], %[r8] \n\t" + "muleq_s.w.phr %[r6], %[r1], %[r2] \n\t" + "muleq_s.w.phr %[r7], %[r0], %[r3] \n\t" + "muleq_s.w.phr %[r8], %[r5], %[r2] \n\t" + "muleq_s.w.phr %[r9], %[r4], %[r3] \n\t" + "addiu %[outre1], %[outre1], 8 \n\t" + "shra_r.w %[r6], %[r6], 6 \n\t" + "shra_r.w %[r7], %[r7], 6 \n\t" + "shra_r.w %[r8], %[r8], 6 \n\t" + "shra_r.w %[r9], %[r9], 6 \n\t" + "subu %[r6], %[r6], %[r7] \n\t" + "addu %[r9], %[r9], %[r8] \n\t" + "subu %[r7], %[r6], %[r9] \n\t" + "addu %[r6], %[r9], %[r6] \n\t" + "negu %[r6], %[r6] \n\t" + "sll %[r10], %[offset], 1 \n\t" + "addu %[r10], %[outre2], %[r10] \n\t" + "sw %[r7], 0(%[outre2]) \n\t" + "absq_s.w %[r7], %[r7] \n\t" + "sw %[r6], 4(%[r10]) \n\t" + "absq_s.w %[r6], %[r6] \n\t" + "slt %[r8], %[max], %[r7] \n\t" + "movn %[max], %[r7], %[r8] \n\t" + "slt %[r8], %[max], %[r6] \n\t" + "movn %[max], %[r6], %[r8] \n\t" + "muleq_s.w.phl %[r6], %[r1], %[r2] \n\t" + "muleq_s.w.phl %[r7], %[r0], %[r3] \n\t" + "muleq_s.w.phl %[r8], %[r5], %[r2] \n\t" + "muleq_s.w.phl %[r9], %[r4], %[r3] \n\t" + "addiu %[offset], %[offset], -8 \n\t" + "shra_r.w %[r6], %[r6], 6 \n\t" + "shra_r.w %[r7], %[r7], 6 \n\t" + "shra_r.w %[r8], %[r8], 6 \n\t" + "shra_r.w %[r9], %[r9], 6 \n\t" + "subu %[r6], %[r6], %[r7] \n\t" + "addu %[r9], %[r9], %[r8] \n\t" + "subu %[r7], %[r6], %[r9] \n\t" + "addu %[r6], %[r9], %[r6] \n\t" + "negu %[r6], %[r6] \n\t" + "sw %[r7], 4(%[outre2]) \n\t" + "absq_s.w %[r7], %[r7] \n\t" + "sw %[r6], 0(%[r10]) \n\t" + "absq_s.w %[r6], %[r6] \n\t" + "slt %[r8], %[max], %[r7] \n\t" + "movn %[max], %[r7], %[r8] \n\t" + "slt %[r8], %[max], %[r6] \n\t" + "movn %[max], %[r6], %[r8] \n\t" + "bgtz %[k], 1b \n\t" + " addiu %[outre2], %[outre2], 8 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "lh %[r0], 0(%[inre]) \n\t" + "lh %[r1], 0(%[inim]) \n\t" + "lh %[r4], 0(%[cosptr]) \n\t" + "lh %[r5], 0(%[sinptr]) \n\t" + "addiu %[k], %[k], -1 \n\t" + "mul %[r2], %[r0], %[r4] \n\t" + "mul %[r0], %[r0], %[r5] \n\t" + "mul %[r3], %[r1], %[r5] \n\t" + "mul %[r1], %[r1], %[r4] \n\t" + "addiu %[cosptr], %[cosptr], 2 \n\t" + "addiu %[sinptr], %[sinptr], 2 \n\t" + "addu %[r8], %[inre], %[offset] \n\t" + "addu %[r9], %[inim], %[offset] \n\t" + "addiu %[r2], %[r2], 16 \n\t" + "sra %[r2], %[r2], 5 \n\t" + "addiu %[r0], %[r0], 16 \n\t" + "sra %[r0], %[r0], 5 \n\t" + "addiu %[r3], %[r3], 16 \n\t" + "sra %[r3], %[r3], 5 \n\t" + "lh %[r6], 0(%[r8]) \n\t" + "lh %[r7], 0(%[r9]) \n\t" + "addiu %[r1], %[r1], 16 \n\t" + "sra %[r1], %[r1], 5 \n\t" + "mul %[r8], %[r7], %[r4] \n\t" + "mul %[r7], %[r7], %[r5] \n\t" + "mul %[r9], %[r6], %[r4] \n\t" + "mul %[r6], %[r6], %[r5] \n\t" + "addu %[r2], %[r2], %[r3] \n\t" + "subu %[r1], %[r1], %[r0] \n\t" + "sll %[r0], %[offset], 1 \n\t" + "addu %[r4], %[outre1], %[r0] \n\t" + "addu %[r5], %[outre2], %[r0] \n\t" + "addiu %[r8], %[r8], 16 \n\t" + "sra %[r8], %[r8], 5 \n\t" + "addiu %[r7], %[r7], 16 \n\t" + "sra %[r7], %[r7], 5 \n\t" + "addiu %[r6], %[r6], 16 \n\t" + "sra %[r6], %[r6], 5 \n\t" + "addiu %[r9], %[r9], 16 \n\t" + "sra %[r9], %[r9], 5 \n\t" + "addu %[r8], %[r8], %[r6] \n\t" + "negu %[r8], %[r8] \n\t" + "subu %[r7], %[r7], %[r9] \n\t" + "subu %[r6], %[r2], %[r7] \n\t" + "addu %[r0], %[r2], %[r7] \n\t" + "addu %[r3], %[r1], %[r8] \n\t" + "subu %[r1], %[r8], %[r1] \n\t" + "sw %[r6], 0(%[outre1]) \n\t" + "sw %[r0], 0(%[r4]) \n\t" + "sw %[r3], 0(%[outre2]) \n\t" + "sw %[r1], 0(%[r5]) \n\t" + "addiu %[outre1], %[outre1], 4 \n\t" + "addiu %[offset], %[offset], -4 \n\t" + "addiu %[inre], %[inre], 2 \n\t" + "addiu %[inim], %[inim], 2 \n\t" + // Inlined WebRtcSpl_MaxAbsValueW32 + "negu %[r5], %[r6] \n\t" + "slt %[r2], %[r6], $zero \n\t" + "movn %[r6], %[r5], %[r2] \n\t" + "negu %[r5], %[r0] \n\t" + "slt %[r2], %[r0], $zero \n\t" + "movn %[r0], %[r5], %[r2] \n\t" + "negu %[r5], %[r3] \n\t" + "slt %[r2], %[r3], $zero \n\t" + "movn %[r3], %[r5], %[r2] \n\t" + "negu %[r5], %[r1] \n\t" + "slt %[r2], %[r1], $zero \n\t" + "movn %[r1], %[r5], %[r2] \n\t" + "slt %[r2], %[r6], %[r0] \n\t" + "slt %[r5], %[r3], %[r1] \n\t" + "movn %[r6], %[r0], %[r2] \n\t" + "movn %[r3], %[r1], %[r5] \n\t" + "slt %[r2], %[r6], %[r3] \n\t" + "movn %[r6], %[r3], %[r2] \n\t" + "slt %[r2], %[max], %[r6] \n\t" + "movn %[max], %[r6], %[r2] \n\t" + "bgtz %[k], 1b \n\t" + " addiu %[outre2], %[outre2], 4 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "clz %[max], %[max] \n\t" + "addiu %[max], %[max], -25 \n\t" + ".set pop \n\t" + : [inre] "=&r" (inre), [inim] "=&r" (inim), + [outre1] "=&r" (outre1), [outre2] "=&r" (outre2), + [offset] "+r" (offset), [k] "+r" (k), [r0] "=&r" (r0), + [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), + [r4] "=&r" (r4), [r5] "=&r" (r5), [r6] "=&r" (r6), + [r7] "=&r" (r7), [r10] "=&r" (r10), + [r8] "=&r" (r8), [r9] "=&r" (r9), [max] "=&r" (max) + : [inreQ7] "r" (inreQ7), [inimQ7] "r" (inimQ7), + [cosptr] "r" (cosptr), [sinptr] "r" (sinptr), + [outre1Q16] "r" (outre1Q16), [outre2Q16] "r" (outre2Q16) + : "hi", "lo", "memory" + ); + + // "Fastest" vectors + k = FRAMESAMPLES / 4; + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[inre], %[inreQ7], 0 \n\t" + "addiu %[inim], %[inimQ7], 0 \n\t" + "addiu %[outre1], %[outre1Q16], 0 \n\t" + "addiu %[outre2], %[outre2Q16], 0 \n\t" + "bltz %[max], 2f \n\t" + " subu %[max1], $zero, %[max] \n\t" + "1: \n\t" + "lw %[r0], 0(%[outre1]) \n\t" + "lw %[r1], 0(%[outre2]) \n\t" + "lw %[r2], 4(%[outre1]) \n\t" + "lw %[r3], 4(%[outre2]) \n\t" + "sllv %[r0], %[r0], %[max] \n\t" + "sllv %[r1], %[r1], %[max] \n\t" + "sllv %[r2], %[r2], %[max] \n\t" + "sllv %[r3], %[r3], %[max] \n\t" + "addiu %[k], %[k], -1 \n\t" + "addiu %[outre1], %[outre1], 8 \n\t" + "addiu %[outre2], %[outre2], 8 \n\t" + "sh %[r0], 0(%[inre]) \n\t" + "sh %[r1], 0(%[inim]) \n\t" + "sh %[r2], 2(%[inre]) \n\t" + "sh %[r3], 2(%[inim]) \n\t" + "addiu %[inre], %[inre], 4 \n\t" + "bgtz %[k], 1b \n\t" + " addiu %[inim], %[inim], 4 \n\t" + "b 4f \n\t" + " nop \n\t" + "2: \n\t" +#if !defined(MIPS_DSP_R1_LE) + "addiu %[r4], $zero, 1 \n\t" + "addiu %[r5], %[max1], -1 \n\t" + "sllv %[r4], %[r4], %[r5] \n\t" +#endif // #if !defined(MIPS_DSP_R1_LE) + "3: \n\t" + "lw %[r0], 0(%[outre1]) \n\t" + "lw %[r1], 0(%[outre2]) \n\t" + "lw %[r2], 4(%[outre1]) \n\t" + "lw %[r3], 4(%[outre2]) \n\t" +#if defined(MIPS_DSP_R1_LE) + "shrav_r.w %[r0], %[r0], %[max1] \n\t" + "shrav_r.w %[r1], %[r1], %[max1] \n\t" + "shrav_r.w %[r2], %[r2], %[max1] \n\t" + "shrav_r.w %[r3], %[r3], %[max1] \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addu %[r0], %[r0], %[r4] \n\t" + "addu %[r1], %[r1], %[r4] \n\t" + "addu %[r2], %[r2], %[r4] \n\t" + "addu %[r3], %[r3], %[r4] \n\t" + "srav %[r0], %[r0], %[max1] \n\t" + "srav %[r1], %[r1], %[max1] \n\t" + "srav %[r2], %[r2], %[max1] \n\t" + "srav %[r3], %[r3], %[max1] \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "addiu %[outre1], %[outre1], 8 \n\t" + "addiu %[outre2], %[outre2], 8 \n\t" + "sh %[r0], 0(%[inre]) \n\t" + "sh %[r1], 0(%[inim]) \n\t" + "sh %[r2], 2(%[inre]) \n\t" + "sh %[r3], 2(%[inim]) \n\t" + "addiu %[k], %[k], -1 \n\t" + "addiu %[inre], %[inre], 4 \n\t" + "bgtz %[k], 3b \n\t" + " addiu %[inim], %[inim], 4 \n\t" + "4: \n\t" + ".set pop \n\t" + : [k] "+r" (k), [max1] "=&r" (max1), [r0] "=&r" (r0), + [inre] "=&r" (inre), [inim] "=&r" (inim), + [outre1] "=&r" (outre1), [outre2] "=&r" (outre2), +#if !defined(MIPS_DSP_R1_LE) + [r4] "=&r" (r4), [r5] "=&r" (r5), +#endif // #if !defined(MIPS_DSP_R1_LE) + [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3) + : [max] "r" (max), [inreQ7] "r" (inreQ7), + [inimQ7] "r" (inimQ7), [outre1Q16] "r" (outre1Q16), + [outre2Q16] "r" (outre2Q16) + : "memory" + ); + + WebRtcIsacfix_FftRadix16Fastest(inreQ7, inimQ7, 1); // real call + + // All the remaining processing is done inside a single loop to avoid + // unnecessary memory accesses. MIPS DSPr2 version processes two samples + // at a time. + cosptr = (int16_t*)WebRtcIsacfix_kCosTab1; + sinptr = (int16_t*)WebRtcIsacfix_kSinTab1; + k = FRAMESAMPLES / 2; + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[inre], %[inreQ7], 0 \n\t" + "addiu %[inim], %[inimQ7], 0 \n\t" + "addiu %[outre1], %[outre1Q16], 0 \n\t" + "addiu %[outre2], %[outre2Q16], 0 \n\t" + "addiu %[r4], $zero, 273 \n\t" + "addiu %[r5], $zero, 31727 \n\t" +#if defined(MIPS_DSP_R2_LE) + "addiu %[max], %[max], 16 \n\t" + "replv.ph %[r4], %[r4] \n\t" +#endif // #if defined(MIPS_DSP_R2_LE) + "bltz %[max], 2f \n\t" + " subu %[max1], $zero, %[max] \n\t" +#if defined(MIPS_DSP_R2_LE) + "addiu %[max], %[max], 1 \n\t" +#endif // #if defined(MIPS_DSP_R2_LE) + "1: \n\t" +#if defined(MIPS_DSP_R2_LE) + "lwl %[r0], 0(%[inre]) \n\t" + "lwl %[r1], 0(%[inim]) \n\t" + "lh %[r2], 0(%[cosptr]) \n\t" + "lwr %[r0], 0(%[inre]) \n\t" + "lwr %[r1], 0(%[inim]) \n\t" + "lh %[r3], 0(%[sinptr]) \n\t" + "muleq_s.w.phr %[r6], %[r0], %[r4] \n\t" + "muleq_s.w.phr %[r7], %[r1], %[r4] \n\t" + "muleq_s.w.phl %[r0], %[r0], %[r4] \n\t" + "muleq_s.w.phl %[r1], %[r1], %[r4] \n\t" + "addiu %[k], %[k], -2 \n\t" + "addiu %[inre], %[inre], 4 \n\t" + "addiu %[inim], %[inim], 4 \n\t" + "shrav_r.w %[r6], %[r6], %[max] \n\t" + "shrav_r.w %[r7], %[r7], %[max] \n\t" + "mult $ac0, %[r2], %[r6] \n\t" + "mult $ac1, %[r3], %[r7] \n\t" + "mult $ac2, %[r2], %[r7] \n\t" + "mult $ac3, %[r3], %[r6] \n\t" + "lh %[r2], 2(%[cosptr]) \n\t" + "lh %[r3], 2(%[sinptr]) \n\t" + "extr_r.w %[r6], $ac0, 14 \n\t" + "extr_r.w %[r7], $ac1, 14 \n\t" + "extr_r.w %[r8], $ac2, 14 \n\t" + "extr_r.w %[r9], $ac3, 14 \n\t" + "shrav_r.w %[r0], %[r0], %[max] \n\t" + "shrav_r.w %[r1], %[r1], %[max] \n\t" + "mult $ac0, %[r2], %[r0] \n\t" + "mult $ac1, %[r3], %[r1] \n\t" + "mult $ac2, %[r2], %[r1] \n\t" + "mult $ac3, %[r3], %[r0] \n\t" + "addiu %[cosptr], %[cosptr], 4 \n\t" + "extr_r.w %[r0], $ac0, 14 \n\t" + "extr_r.w %[r1], $ac1, 14 \n\t" + "extr_r.w %[r2], $ac2, 14 \n\t" + "extr_r.w %[r3], $ac3, 14 \n\t" + "subu %[r6], %[r6], %[r7] \n\t" + "addu %[r8], %[r8], %[r9] \n\t" + "mult $ac0, %[r5], %[r6] \n\t" + "mult $ac1, %[r5], %[r8] \n\t" + "addiu %[sinptr], %[sinptr], 4 \n\t" + "subu %[r0], %[r0], %[r1] \n\t" + "addu %[r2], %[r2], %[r3] \n\t" + "extr_r.w %[r1], $ac0, 11 \n\t" + "extr_r.w %[r3], $ac1, 11 \n\t" + "mult $ac2, %[r5], %[r0] \n\t" + "mult $ac3, %[r5], %[r2] \n\t" + "sw %[r1], 0(%[outre1]) \n\t" + "sw %[r3], 0(%[outre2]) \n\t" + "addiu %[outre1], %[outre1], 8 \n\t" + "extr_r.w %[r0], $ac2, 11 \n\t" + "extr_r.w %[r2], $ac3, 11 \n\t" + "sw %[r0], -4(%[outre1]) \n\t" + "sw %[r2], 4(%[outre2]) \n\t" + "bgtz %[k], 1b \n\t" + " addiu %[outre2], %[outre2], 8 \n\t" + "b 3f \n\t" +#else // #if defined(MIPS_DSP_R2_LE) + "lh %[r0], 0(%[inre]) \n\t" + "lh %[r1], 0(%[inim]) \n\t" + "addiu %[k], %[k], -1 \n\t" + "srav %[r0], %[r0], %[max] \n\t" + "srav %[r1], %[r1], %[max] \n\t" + "sra %[r2], %[r0], 16 \n\t" + "andi %[r0], %[r0], 0xFFFF \n\t" + "sra %[r0], %[r0], 1 \n\t" + "sra %[r3], %[r1], 16 \n\t" + "andi %[r1], %[r1], 0xFFFF \n\t" + "sra %[r1], %[r1], 1 \n\t" + "mul %[r2], %[r2], %[r4] \n\t" + "mul %[r0], %[r0], %[r4] \n\t" + "mul %[r3], %[r3], %[r4] \n\t" + "mul %[r1], %[r1], %[r4] \n\t" + "addiu %[inre], %[inre], 2 \n\t" + "addiu %[inim], %[inim], 2 \n\t" + "lh %[r6], 0(%[cosptr]) \n\t" + "lh %[r7], 0(%[sinptr]) \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r0], %[r0], 15 \n\t" + "shra_r.w %[r1], %[r1], 15 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r0], %[r0], 0x4000 \n\t" + "addiu %[r1], %[r1], 0x4000 \n\t" + "sra %[r0], %[r0], 15 \n\t" + "sra %[r1], %[r1], 15 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "addu %[r0], %[r2], %[r0] \n\t" + "addu %[r1], %[r3], %[r1] \n\t" + "sra %[r2], %[r0], 16 \n\t" + "andi %[r0], %[r0], 0xFFFF \n\t" + "mul %[r9], %[r2], %[r6] \n\t" + "mul %[r2], %[r2], %[r7] \n\t" + "mul %[r8], %[r0], %[r6] \n\t" + "mul %[r0], %[r0], %[r7] \n\t" + "sra %[r3], %[r3], 16 \n\t" + "andi %[r1], %[r1], 0xFFFF \n\t" + "sll %[r9], %[r9], 2 \n\t" + "sll %[r2], %[r2], 2 \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r8], %[r8], 14 \n\t" + "shra_r.w %[r0], %[r0], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r8], %[r8], 0x2000 \n\t" + "addiu %[r0], %[r0], 0x2000 \n\t" + "sra %[r8], %[r8], 14 \n\t" + "sra %[r0], %[r0], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "addu %[r9], %[r9], %[r8] \n\t" + "addu %[r2], %[r2], %[r0] \n\t" + "mul %[r0], %[r3], %[r6] \n\t" + "mul %[r3], %[r3], %[r7] \n\t" + "mul %[r8], %[r1], %[r6] \n\t" + "mul %[r1], %[r1], %[r8] \n\t" + "addiu %[cosptr], %[cosptr], 2 \n\t" + "addiu %[sinptr], %[sinptr], 2 \n\t" + "sll %[r0], %[r0], 2 \n\t" + "sll %[r3], %[r3], 2 \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r8], %[r8], 14 \n\t" + "shra_r.w %[r1], %[r1], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r8], %[r8], 0x2000 \n\t" + "addiu %[r1], %[r1], 0x2000 \n\t" + "sra %[r8], %[r8], 14 \n\t" + "sra %[r1], %[r1], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "addu %[r0], %[r0], %[r8] \n\t" + "addu %[r3], %[r3], %[r1] \n\t" + "subu %[r9], %[r9], %[r3] \n\t" + "addu %[r0], %[r0], %[r2] \n\t" + "sra %[r1], %[r9], 16 \n\t" + "andi %[r9], %[r9], 0xFFFF \n\t" + "mul %[r1], %[r1], %[r5] \n\t" + "mul %[r9], %[r9], %[r5] \n\t" + "sra %[r2], %[r0], 16 \n\t" + "andi %[r0], %[r0], 0xFFFF \n\t" + "mul %[r2], %[r2], %[r5] \n\t" + "mul %[r0], %[r0], %[r5] \n\t" + "sll %[r1], %[r1], 5 \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r9], %[r9], 11 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r9], %[r9], 0x400 \n\t" + "sra %[r9], %[r9], 11 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "addu %[r1], %[r1], %[r9] \n\t" + "sll %[r2], %[r2], 5 \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r0], %[r0], 11 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r0], %[r0], 0x400 \n\t" + "sra %[r0], %[r0], 11 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "addu %[r0], %[r0], %[r2] \n\t" + "sw %[r1], 0(%[outre1]) \n\t" + "addiu %[outre1], %[outre1], 4 \n\t" + "sw %[r0], 0(%[outre2]) \n\t" + "bgtz %[k], 1b \n\t" + " addiu %[outre2], %[outre2], 4 \n\t" + "b 3f \n\t" + " nop \n\t" +#endif // #if defined(MIPS_DSP_R2_LE) + "2: \n\t" +#if defined(MIPS_DSP_R2_LE) + "addiu %[max1], %[max1], -1 \n\t" + "21: \n\t" + "lwl %[r0], 0(%[inre]) \n\t" + "lwl %[r1], 0(%[inim]) \n\t" + "lh %[r2], 0(%[cosptr]) \n\t" + "lwr %[r0], 0(%[inre]) \n\t" + "lwr %[r1], 0(%[inim]) \n\t" + "lh %[r3], 0(%[sinptr]) \n\t" + "muleq_s.w.phr %[r6], %[r0], %[r4] \n\t" + "muleq_s.w.phr %[r7], %[r1], %[r4] \n\t" + "muleq_s.w.phl %[r0], %[r0], %[r4] \n\t" + "muleq_s.w.phl %[r1], %[r1], %[r4] \n\t" + "addiu %[k], %[k], -2 \n\t" + "addiu %[inre], %[inre], 4 \n\t" + "addiu %[inim], %[inim], 4 \n\t" + "sllv %[r6], %[r6], %[max1] \n\t" + "sllv %[r7], %[r7], %[max1] \n\t" + "mult $ac0, %[r2], %[r6] \n\t" + "mult $ac1, %[r3], %[r7] \n\t" + "mult $ac2, %[r2], %[r7] \n\t" + "mult $ac3, %[r3], %[r6] \n\t" + "lh %[r2], 2(%[cosptr]) \n\t" + "lh %[r3], 2(%[sinptr]) \n\t" + "extr_r.w %[r6], $ac0, 14 \n\t" + "extr_r.w %[r7], $ac1, 14 \n\t" + "extr_r.w %[r8], $ac2, 14 \n\t" + "extr_r.w %[r9], $ac3, 14 \n\t" + "sllv %[r0], %[r0], %[max1] \n\t" + "sllv %[r1], %[r1], %[max1] \n\t" + "mult $ac0, %[r2], %[r0] \n\t" + "mult $ac1, %[r3], %[r1] \n\t" + "mult $ac2, %[r2], %[r1] \n\t" + "mult $ac3, %[r3], %[r0] \n\t" + "addiu %[cosptr], %[cosptr], 4 \n\t" + "extr_r.w %[r0], $ac0, 14 \n\t" + "extr_r.w %[r1], $ac1, 14 \n\t" + "extr_r.w %[r2], $ac2, 14 \n\t" + "extr_r.w %[r3], $ac3, 14 \n\t" + "subu %[r6], %[r6], %[r7] \n\t" + "addu %[r8], %[r8], %[r9] \n\t" + "mult $ac0, %[r5], %[r6] \n\t" + "mult $ac1, %[r5], %[r8] \n\t" + "addiu %[sinptr], %[sinptr], 4 \n\t" + "subu %[r0], %[r0], %[r1] \n\t" + "addu %[r2], %[r2], %[r3] \n\t" + "extr_r.w %[r1], $ac0, 11 \n\t" + "extr_r.w %[r3], $ac1, 11 \n\t" + "mult $ac2, %[r5], %[r0] \n\t" + "mult $ac3, %[r5], %[r2] \n\t" + "sw %[r1], 0(%[outre1]) \n\t" + "sw %[r3], 0(%[outre2]) \n\t" + "addiu %[outre1], %[outre1], 8 \n\t" + "extr_r.w %[r0], $ac2, 11 \n\t" + "extr_r.w %[r2], $ac3, 11 \n\t" + "sw %[r0], -4(%[outre1]) \n\t" + "sw %[r2], 4(%[outre2]) \n\t" + "bgtz %[k], 21b \n\t" + " addiu %[outre2], %[outre2], 8 \n\t" + "b 3f \n\t" + " nop \n\t" +#else // #if defined(MIPS_DSP_R2_LE) + "lh %[r0], 0(%[inre]) \n\t" + "lh %[r1], 0(%[inim]) \n\t" + "addiu %[k], %[k], -1 \n\t" + "sllv %[r0], %[r0], %[max1] \n\t" + "sllv %[r1], %[r1], %[max1] \n\t" + "sra %[r2], %[r0], 16 \n\t" + "andi %[r0], %[r0], 0xFFFF \n\t" + "sra %[r0], %[r0], 1 \n\t" + "sra %[r3], %[r1], 16 \n\t" + "andi %[r1], %[r1], 0xFFFF \n\t" + "sra %[r1], %[r1], 1 \n\t" + "mul %[r2], %[r2], %[r4] \n\t" + "mul %[r0], %[r0], %[r4] \n\t" + "mul %[r3], %[r3], %[r4] \n\t" + "mul %[r1], %[r1], %[r4] \n\t" + "addiu %[inre], %[inre], 2 \n\t" + "addiu %[inim], %[inim], 2 \n\t" + "lh %[r6], 0(%[cosptr]) \n\t" + "lh %[r7], 0(%[sinptr]) \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r0], %[r0], 15 \n\t" + "shra_r.w %[r1], %[r1], 15 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r0], %[r0], 0x4000 \n\t" + "addiu %[r1], %[r1], 0x4000 \n\t" + "sra %[r0], %[r0], 15 \n\t" + "sra %[r1], %[r1], 15 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "addu %[r0], %[r2], %[r0] \n\t" + "addu %[r1], %[r3], %[r1] \n\t" + "sra %[r2], %[r0], 16 \n\t" + "andi %[r0], %[r0], 0xFFFF \n\t" + "mul %[r9], %[r2], %[r6] \n\t" + "mul %[r2], %[r2], %[r7] \n\t" + "mul %[r8], %[r0], %[r6] \n\t" + "mul %[r0], %[r0], %[r7] \n\t" + "sra %[r3], %[r1], 16 \n\t" + "andi %[r1], %[r1], 0xFFFF \n\t" + "sll %[r9], %[r9], 2 \n\t" + "sll %[r2], %[r2], 2 \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r8], %[r8], 14 \n\t" + "shra_r.w %[r0], %[r0], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r8], %[r8], 0x2000 \n\t" + "addiu %[r0], %[r0], 0x2000 \n\t" + "sra %[r8], %[r8], 14 \n\t" + "sra %[r0], %[r0], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "addu %[r9], %[r9], %[r8] \n\t" + "addu %[r2], %[r2], %[r0] \n\t" + "mul %[r0], %[r3], %[r6] \n\t" + "mul %[r3], %[r3], %[r7] \n\t" + "mul %[r8], %[r1], %[r6] \n\t" + "mul %[r1], %[r1], %[r7] \n\t" + "addiu %[cosptr], %[cosptr], 2 \n\t" + "addiu %[sinptr], %[sinptr], 2 \n\t" + "sll %[r0], %[r0], 2 \n\t" + "sll %[r3], %[r3], 2 \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r8], %[r8], 14 \n\t" + "shra_r.w %[r1], %[r1], 14 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r8], %[r8], 0x2000 \n\t" + "addiu %[r1], %[r1], 0x2000 \n\t" + "sra %[r8], %[r8], 14 \n\t" + "sra %[r1], %[r1], 14 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "addu %[r0], %[r0], %[r8] \n\t" + "addu %[r3], %[r3], %[r1] \n\t" + "subu %[r9], %[r9], %[r3] \n\t" + "addu %[r0], %[r0], %[r2] \n\t" + "sra %[r1], %[r9], 16 \n\t" + "andi %[r9], %[r9], 0xFFFF \n\t" + "mul %[r1], %[r1], %[r5] \n\t" + "mul %[r9], %[r9], %[r5] \n\t" + "sra %[r2], %[r0], 16 \n\t" + "andi %[r0], %[r0], 0xFFFF \n\t" + "mul %[r2], %[r2], %[r5] \n\t" + "mul %[r0], %[r0], %[r5] \n\t" + "sll %[r1], %[r1], 5 \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r9], %[r9], 11 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r9], %[r9], 0x400 \n\t" + "sra %[r9], %[r9], 11 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "addu %[r1], %[r1], %[r9] \n\t" + "sll %[r2], %[r2], 5 \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[r0], %[r0], 11 \n\t" +#else // #if defined(MIPS_DSP_R1_LE) + "addiu %[r0], %[r0], 0x400 \n\t" + "sra %[r0], %[r0], 11 \n\t" +#endif // #if defined(MIPS_DSP_R1_LE) + "addu %[r0], %[r0], %[r2] \n\t" + "sw %[r1], 0(%[outre1]) \n\t" + "addiu %[outre1], %[outre1], 4 \n\t" + "sw %[r0], 0(%[outre2]) \n\t" + "bgtz %[k], 2b \n\t" + " addiu %[outre2], %[outre2], 4 \n\t" +#endif // #if defined(MIPS_DSP_R2_LE) + "3: \n\t" + ".set pop \n\t" + : [k] "+r" (k), [r0] "=&r" (r0), [r1] "=&r" (r1), + [r2] "=&r" (r2), [r3] "=&r" (r3), [r4] "=&r" (r4), + [r5] "=&r" (r5), [r6] "=&r" (r6), [r7] "=&r" (r7), + [r8] "=&r" (r8), [r9] "=&r" (r9), [max1] "=&r" (max1), + [inre] "=&r" (inre), [inim] "=&r" (inim), + [outre1] "=&r" (outre1), [outre2] "=&r" (outre2) + : [max] "r" (max), [inreQ7] "r" (inreQ7), + [inimQ7] "r" (inimQ7), [cosptr] "r" (cosptr), + [sinptr] "r" (sinptr), [outre1Q16] "r" (outre1Q16), + [outre2Q16] "r" (outre2Q16) + : "hi", "lo", "memory" +#if defined(MIPS_DSP_R2_LE) + , "$ac1hi", "$ac1lo", "$ac2hi", "$ac2lo", "$ac3hi", "$ac3lo" +#endif // #if defined(MIPS_DSP_R2_LE) + ); +} diff --git a/modules/audio_coding/codecs/isac/fix/source/transform_tables.c b/modules/audio_coding/codecs/isac/fix/source/transform_tables.c new file mode 100644 index 00000000..ee96b8e3 --- /dev/null +++ b/modules/audio_coding/codecs/isac/fix/source/transform_tables.c @@ -0,0 +1,111 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/* + * This file contains trigonometric functions look-up tables used in + * transform functions WebRtcIsacfix_Time2Spec and WebRtcIsacfix_Spec2Time. + */ + +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h" +#include "webrtc/typedefs.h" + +#if !(defined WEBRTC_DETECT_ARM_NEON || defined WEBRTC_ARCH_ARM_NEON) +/* Cosine table 1 in Q14. */ +const int16_t WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2] = { + 16384, 16383, 16378, 16371, 16362, 16349, 16333, 16315, 16294, 16270, + 16244, 16214, 16182, 16147, 16110, 16069, 16026, 15980, 15931, 15880, + 15826, 15769, 15709, 15647, 15582, 15515, 15444, 15371, 15296, 15218, + 15137, 15053, 14968, 14879, 14788, 14694, 14598, 14500, 14399, 14295, + 14189, 14081, 13970, 13856, 13741, 13623, 13502, 13380, 13255, 13128, + 12998, 12867, 12733, 12597, 12458, 12318, 12176, 12031, 11885, 11736, + 11585, 11433, 11278, 11121, 10963, 10803, 10641, 10477, 10311, 10143, + 9974, 9803, 9630, 9456, 9280, 9102, 8923, 8743, 8561, 8377, + 8192, 8006, 7818, 7629, 7438, 7246, 7053, 6859, 6664, 6467, + 6270, 6071, 5872, 5671, 5469, 5266, 5063, 4859, 4653, 4447, + 4240, 4033, 3825, 3616, 3406, 3196, 2986, 2775, 2563, 2351, + 2139, 1926, 1713, 1499, 1285, 1072, 857, 643, 429, 214, + 0, -214, -429, -643, -857, -1072, -1285, -1499, -1713, -1926, + -2139, -2351, -2563, -2775, -2986, -3196, -3406, -3616, -3825, -4033, + -4240, -4447, -4653, -4859, -5063, -5266, -5469, -5671, -5872, -6071, + -6270, -6467, -6664, -6859, -7053, -7246, -7438, -7629, -7818, -8006, + -8192, -8377, -8561, -8743, -8923, -9102, -9280, -9456, -9630, -9803, + -9974, -10143, -10311, -10477, -10641, -10803, -10963, -11121, -11278, -11433, + -11585, -11736, -11885, -12031, -12176, -12318, -12458, -12597, -12733, + -12867, -12998, -13128, -13255, -13380, -13502, -13623, -13741, -13856, + -13970, -14081, -14189, -14295, -14399, -14500, -14598, -14694, -14788, + -14879, -14968, -15053, -15137, -15218, -15296, -15371, -15444, -15515, + -15582, -15647, -15709, -15769, -15826, -15880, -15931, -15980, -16026, + -16069, -16110, -16147, -16182, -16214, -16244, -16270, -16294, -16315, + -16333, -16349, -16362, -16371, -16378, -16383 +}; + +/* Sine table 1 in Q14. */ +const int16_t WebRtcIsacfix_kSinTab1[FRAMESAMPLES/2] = { + 0, 214, 429, 643, 857, 1072, 1285, 1499, 1713, 1926, + 2139, 2351, 2563, 2775, 2986, 3196, 3406, 3616, 3825, 4033, + 4240, 4447, 4653, 4859, 5063, 5266, 5469, 5671, 5872, 6071, + 6270, 6467, 6664, 6859, 7053, 7246, 7438, 7629, 7818, 8006, + 8192, 8377, 8561, 8743, 8923, 9102, 9280, 9456, 9630, 9803, + 9974, 10143, 10311, 10477, 10641, 10803, 10963, 11121, 11278, 11433, + 11585, 11736, 11885, 12031, 12176, 12318, 12458, 12597, 12733, 12867, + 12998, 13128, 13255, 13380, 13502, 13623, 13741, 13856, 13970, 14081, + 14189, 14295, 14399, 14500, 14598, 14694, 14788, 14879, 14968, 15053, + 15137, 15218, 15296, 15371, 15444, 15515, 15582, 15647, 15709, 15769, + 15826, 15880, 15931, 15980, 16026, 16069, 16110, 16147, 16182, 16214, + 16244, 16270, 16294, 16315, 16333, 16349, 16362, 16371, 16378, 16383, + 16384, 16383, 16378, 16371, 16362, 16349, 16333, 16315, 16294, 16270, + 16244, 16214, 16182, 16147, 16110, 16069, 16026, 15980, 15931, 15880, + 15826, 15769, 15709, 15647, 15582, 15515, 15444, 15371, 15296, 15218, + 15137, 15053, 14968, 14879, 14788, 14694, 14598, 14500, 14399, 14295, + 14189, 14081, 13970, 13856, 13741, 13623, 13502, 13380, 13255, 13128, + 12998, 12867, 12733, 12597, 12458, 12318, 12176, 12031, 11885, 11736, + 11585, 11433, 11278, 11121, 10963, 10803, 10641, 10477, 10311, 10143, + 9974, 9803, 9630, 9456, 9280, 9102, 8923, 8743, 8561, 8377, + 8192, 8006, 7818, 7629, 7438, 7246, 7053, 6859, 6664, 6467, + 6270, 6071, 5872, 5671, 5469, 5266, 5063, 4859, 4653, 4447, + 4240, 4033, 3825, 3616, 3406, 3196, 2986, 2775, 2563, 2351, + 2139, 1926, 1713, 1499, 1285, 1072, 857, 643, 429, 214 +}; + + +/* Sine table 2 in Q14. */ +const int16_t WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4] = { + 16384, -16381, 16375, -16367, 16356, -16342, 16325, -16305, 16283, -16257, + 16229, -16199, 16165, -16129, 16090, -16048, 16003, -15956, 15906, -15853, + 15798, -15739, 15679, -15615, 15549, -15480, 15408, -15334, 15257, -15178, + 15095, -15011, 14924, -14834, 14741, -14647, 14549, -14449, 14347, -14242, + 14135, -14025, 13913, -13799, 13682, -13563, 13441, -13318, 13192, -13063, + 12933, -12800, 12665, -12528, 12389, -12247, 12104, -11958, 11810, -11661, + 11509, -11356, 11200, -11042, 10883, -10722, 10559, -10394, 10227, -10059, + 9889, -9717, 9543, -9368, 9191, -9013, 8833, -8652, 8469, -8285, + 8099, -7912, 7723, -7534, 7342, -7150, 6957, -6762, 6566, -6369, + 6171, -5971, 5771, -5570, 5368, -5165, 4961, -4756, 4550, -4344, + 4137, -3929, 3720, -3511, 3301, -3091, 2880, -2669, 2457, -2245, + 2032, -1819, 1606, -1392, 1179, -965, 750, -536, 322, -107 +}; +#endif + +#if defined(MIPS32_LE) +/* Cosine table 2 in Q14. Used only on MIPS platforms. */ +const int16_t WebRtcIsacfix_kCosTab2[FRAMESAMPLES/4] = { + 107, -322, 536, -750, 965, -1179, 1392, -1606, 1819, -2032, + 2245, -2457, 2669, -2880, 3091, -3301, 3511, -3720, 3929, -4137, + 4344, -4550, 4756, -4961, 5165, -5368, 5570, -5771, 5971, -6171, + 6369, -6566, 6762, -6957, 7150, -7342, 7534, -7723, 7912, -8099, + 8285, -8469, 8652, -8833, 9013, -9191, 9368, -9543, 9717, -9889, + 10059, -10227, 10394, -10559, 10722, -10883, 11042, -11200, 11356, -11509, + 11661, -11810, 11958, -12104, 12247, -12389, 12528, -12665, 12800, -12933, + 13063, -13192, 13318, -13441, 13563, -13682, 13799, -13913, 14025, -14135, + 14242, -14347, 14449, -14549, 14647, -14741, 14834, -14924, 15011, -15095, + 15178, -15257, 15334, -15408, 15480, -15549, 15615, -15679, 15739, -15798, + 15853, -15906, 15956, -16003, 16048, -16090, 16129, -16165, 16199, -16229, + 16257, -16283, 16305, -16325, 16342, -16356, 16367, -16375, 16381, -16384 +}; +#endif diff --git a/modules/audio_coding/codecs/opus/opus_fec_test.cc b/modules/audio_coding/codecs/opus/opus_fec_test.cc index fb4cb04f..ee027e80 100644 --- a/modules/audio_coding/codecs/opus/opus_fec_test.cc +++ b/modules/audio_coding/codecs/opus/opus_fec_test.cc @@ -32,8 +32,7 @@ struct mode { }; const int kOpusBlockDurationMs = 20; -const int kOpusInputSamplingKhz = 48; -const int kOpusOutputSamplingKhz = 32; +const int kOpusSamplingKhz = 48; class OpusFecTest : public TestWithParam<coding_param> { protected: @@ -47,14 +46,8 @@ class OpusFecTest : public TestWithParam<coding_param> { virtual void DecodeABlock(bool lost_previous, bool lost_current); int block_duration_ms_; - int input_sampling_khz_; - int output_sampling_khz_; - - // Number of samples-per-channel in a frame. - int input_length_sample_; - - // Expected output number of samples-per-channel in a frame. - int output_length_sample_; + int sampling_khz_; + int block_length_sample_; int channels_; int bit_rate_; @@ -91,7 +84,7 @@ void OpusFecTest::SetUp() { // Allocate memory to contain the whole file. in_data_.reset(new int16_t[loop_length_samples_ + - input_length_sample_ * channels_]); + block_length_sample_ * channels_]); // Copy the file into the buffer. ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp), @@ -104,12 +97,12 @@ void OpusFecTest::SetUp() { // beginning of the array. Audio frames cross the end of the excerpt always // appear as a continuum of memory. memcpy(&in_data_[loop_length_samples_], &in_data_[0], - input_length_sample_ * channels_ * sizeof(int16_t)); + block_length_sample_ * channels_ * sizeof(int16_t)); // Maximum number of bytes in output bitstream. - max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t); + max_bytes_ = block_length_sample_ * channels_ * sizeof(int16_t); - out_data_.reset(new int16_t[2 * output_length_sample_ * channels_]); + out_data_.reset(new int16_t[2 * block_length_sample_ * channels_]); bit_stream_.reset(new uint8_t[max_bytes_]); // Create encoder memory. @@ -127,10 +120,8 @@ void OpusFecTest::TearDown() { OpusFecTest::OpusFecTest() : block_duration_ms_(kOpusBlockDurationMs), - input_sampling_khz_(kOpusInputSamplingKhz), - output_sampling_khz_(kOpusOutputSamplingKhz), - input_length_sample_(block_duration_ms_ * input_sampling_khz_), - output_length_sample_(block_duration_ms_ * output_sampling_khz_), + sampling_khz_(kOpusSamplingKhz), + block_length_sample_(block_duration_ms_ * sampling_khz_), data_pointer_(0), max_bytes_(0), encoded_bytes_(0), @@ -141,7 +132,7 @@ OpusFecTest::OpusFecTest() void OpusFecTest::EncodeABlock() { int16_t value = WebRtcOpus_Encode(opus_encoder_, &in_data_[data_pointer_], - input_length_sample_, + block_length_sample_, max_bytes_, &bit_stream_[0]); EXPECT_GT(value, 0); @@ -162,7 +153,7 @@ void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) { } else { value_1 = WebRtcOpus_DecodePlc(opus_decoder_, &out_data_[0], 1); } - EXPECT_EQ(output_length_sample_, value_1); + EXPECT_EQ(block_length_sample_, value_1); } if (!lost_current) { @@ -171,7 +162,7 @@ void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) { encoded_bytes_, &out_data_[value_1 * channels_], &audio_type); - EXPECT_EQ(output_length_sample_, value_2); + EXPECT_EQ(block_length_sample_, value_2); } } @@ -224,7 +215,7 @@ TEST_P(OpusFecTest, RandomPacketLossTest) { // |data_pointer_| is incremented and wrapped across // |loop_length_samples_|. - data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) % + data_pointer_ = (data_pointer_ + block_length_sample_ * channels_) % loop_length_samples_; } if (mode_set[i].fec) { diff --git a/modules/audio_coding/codecs/opus/opus_interface.c b/modules/audio_coding/codecs/opus/opus_interface.c index 24fc4fc4..ea535ea9 100644 --- a/modules/audio_coding/codecs/opus/opus_interface.c +++ b/modules/audio_coding/codecs/opus/opus_interface.c @@ -15,9 +15,6 @@ #include "opus.h" -#include "webrtc/common_audio/signal_processing/resample_by_2_internal.h" -#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" - enum { /* Maximum supported frame size in WebRTC is 60 ms. */ kWebRtcOpusMaxEncodeFrameSizeMs = 60, @@ -31,17 +28,6 @@ enum { * milliseconds. */ kWebRtcOpusMaxFrameSizePerChannel = 48 * kWebRtcOpusMaxDecodeFrameSizeMs, - /* Maximum sample count per frame is 48 kHz * maximum frame size in - * milliseconds * maximum number of channels. */ - kWebRtcOpusMaxFrameSize = kWebRtcOpusMaxFrameSizePerChannel * 2, - - /* Maximum sample count per channel for output resampled to 32 kHz, - * 32 kHz * maximum frame size in milliseconds. */ - kWebRtcOpusMaxFrameSizePerChannel32kHz = 32 * kWebRtcOpusMaxDecodeFrameSizeMs, - - /* Number of samples in resampler state. */ - kWebRtcOpusStateSize = 7, - /* Default frame size, 20 ms @ 48 kHz, in samples (for one channel). */ kWebRtcOpusDefaultFrameSize = 960, }; @@ -143,8 +129,6 @@ int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) { } struct WebRtcOpusDecInst { - int16_t state_48_32_left[8]; - int16_t state_48_32_right[8]; OpusDecoder* decoder_left; OpusDecoder* decoder_right; int prev_decoded_samples; @@ -205,8 +189,6 @@ int WebRtcOpus_DecoderChannels(OpusDecInst* inst) { int16_t WebRtcOpus_DecoderInitNew(OpusDecInst* inst) { int error = opus_decoder_ctl(inst->decoder_left, OPUS_RESET_STATE); if (error == OPUS_OK) { - memset(inst->state_48_32_left, 0, sizeof(inst->state_48_32_left)); - memset(inst->state_48_32_right, 0, sizeof(inst->state_48_32_right)); return 0; } return -1; @@ -215,7 +197,6 @@ int16_t WebRtcOpus_DecoderInitNew(OpusDecInst* inst) { int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) { int error = opus_decoder_ctl(inst->decoder_left, OPUS_RESET_STATE); if (error == OPUS_OK) { - memset(inst->state_48_32_left, 0, sizeof(inst->state_48_32_left)); return 0; } return -1; @@ -224,7 +205,6 @@ int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) { int16_t WebRtcOpus_DecoderInitSlave(OpusDecInst* inst) { int error = opus_decoder_ctl(inst->decoder_right, OPUS_RESET_STATE); if (error == OPUS_OK) { - memset(inst->state_48_32_right, 0, sizeof(inst->state_48_32_right)); return 0; } return -1; @@ -267,124 +247,29 @@ static int DecodeFec(OpusDecoder* inst, const int16_t* encoded, return -1; } -/* Resample from 48 to 32 kHz. Length of state is assumed to be - * kWebRtcOpusStateSize (7). - */ -static int WebRtcOpus_Resample48to32(const int16_t* samples_in, int length, - int16_t* state, int16_t* samples_out) { - int i; - int blocks; - int16_t output_samples; - int32_t buffer32[kWebRtcOpusMaxFrameSizePerChannel + kWebRtcOpusStateSize]; - - /* Resample from 48 kHz to 32 kHz. */ - for (i = 0; i < kWebRtcOpusStateSize; i++) { - buffer32[i] = state[i]; - state[i] = samples_in[length - kWebRtcOpusStateSize + i]; - } - for (i = 0; i < length; i++) { - buffer32[kWebRtcOpusStateSize + i] = samples_in[i]; - } - /* Resampling 3 samples to 2. Function divides the input in |blocks| number - * of 3-sample groups, and output is |blocks| number of 2-sample groups. - * When this is removed, the compensation in WebRtcOpus_DurationEst should be - * removed too. */ - blocks = length / 3; - WebRtcSpl_Resample48khzTo32khz(buffer32, buffer32, blocks); - output_samples = (int16_t) (blocks * 2); - WebRtcSpl_VectorBitShiftW32ToW16(samples_out, output_samples, buffer32, 15); - - return output_samples; -} - -static int WebRtcOpus_DeInterleaveResample(OpusDecInst* inst, int16_t* input, - int sample_pairs, int16_t* output) { - int i; - int16_t buffer_left[kWebRtcOpusMaxFrameSizePerChannel]; - int16_t buffer_right[kWebRtcOpusMaxFrameSizePerChannel]; - int16_t buffer_out[kWebRtcOpusMaxFrameSizePerChannel32kHz]; - int resampled_samples; - - /* De-interleave the signal in left and right channel. */ - for (i = 0; i < sample_pairs; i++) { - /* Take every second sample, starting at the first sample. */ - buffer_left[i] = input[i * 2]; - buffer_right[i] = input[i * 2 + 1]; - } - - /* Resample from 48 kHz to 32 kHz for left channel. */ - resampled_samples = WebRtcOpus_Resample48to32( - buffer_left, sample_pairs, inst->state_48_32_left, buffer_out); - - /* Add samples interleaved to output vector. */ - for (i = 0; i < resampled_samples; i++) { - output[i * 2] = buffer_out[i]; - } - - /* Resample from 48 kHz to 32 kHz for right channel. */ - resampled_samples = WebRtcOpus_Resample48to32( - buffer_right, sample_pairs, inst->state_48_32_right, buffer_out); - - /* Add samples interleaved to output vector. */ - for (i = 0; i < resampled_samples; i++) { - output[i * 2 + 1] = buffer_out[i]; - } - - return resampled_samples; -} - int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded, int16_t encoded_bytes, int16_t* decoded, int16_t* audio_type) { - /* |buffer| is big enough for 120 ms (the largest Opus packet size) of stereo - * audio at 48 kHz. */ - int16_t buffer[kWebRtcOpusMaxFrameSize]; int16_t* coded = (int16_t*)encoded; int decoded_samples; - int resampled_samples; - - /* If mono case, just do a regular call to the decoder. - * If stereo, we need to de-interleave the stereo output into blocks with - * left and right channel. Each block is resampled to 32 kHz, and then - * interleaved again. */ - /* Decode to a temporary buffer. */ decoded_samples = DecodeNative(inst->decoder_left, coded, encoded_bytes, kWebRtcOpusMaxFrameSizePerChannel, - buffer, audio_type); + decoded, audio_type); if (decoded_samples < 0) { return -1; } - if (inst->channels == 2) { - /* De-interleave and resample. */ - resampled_samples = WebRtcOpus_DeInterleaveResample(inst, - buffer, - decoded_samples, - decoded); - } else { - /* Resample from 48 kHz to 32 kHz. Filter state memory for left channel is - * used for mono signals. */ - resampled_samples = WebRtcOpus_Resample48to32(buffer, - decoded_samples, - inst->state_48_32_left, - decoded); - } - /* Update decoded sample memory, to be used by the PLC in case of losses. */ inst->prev_decoded_samples = decoded_samples; - return resampled_samples; + return decoded_samples; } int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded, int16_t encoded_bytes, int16_t* decoded, int16_t* audio_type) { - /* |buffer16| is big enough for 120 ms (the largestOpus packet size) of - * stereo audio at 48 kHz. */ - int16_t buffer16[kWebRtcOpusMaxFrameSize]; int decoded_samples; - int16_t output_samples; int i; /* If mono case, just do a regular call to the decoder. @@ -393,120 +278,82 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded, * This is to make stereo work with the current setup of NetEQ, which * requires two calls to the decoder to produce stereo. */ - /* Decode to a temporary buffer. */ decoded_samples = DecodeNative(inst->decoder_left, encoded, encoded_bytes, - kWebRtcOpusMaxFrameSizePerChannel, buffer16, + kWebRtcOpusMaxFrameSizePerChannel, decoded, audio_type); if (decoded_samples < 0) { return -1; } if (inst->channels == 2) { /* The parameter |decoded_samples| holds the number of samples pairs, in - * case of stereo. Number of samples in |buffer16| equals |decoded_samples| + * case of stereo. Number of samples in |decoded| equals |decoded_samples| * times 2. */ for (i = 0; i < decoded_samples; i++) { /* Take every second sample, starting at the first sample. This gives * the left channel. */ - buffer16[i] = buffer16[i * 2]; + decoded[i] = decoded[i * 2]; } } - /* Resample from 48 kHz to 32 kHz. */ - output_samples = WebRtcOpus_Resample48to32(buffer16, decoded_samples, - inst->state_48_32_left, decoded); - /* Update decoded sample memory, to be used by the PLC in case of losses. */ inst->prev_decoded_samples = decoded_samples; - return output_samples; + return decoded_samples; } int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const int16_t* encoded, int16_t encoded_bytes, int16_t* decoded, int16_t* audio_type) { - /* |buffer16| is big enough for 120 ms (the largestOpus packet size) of - * stereo audio at 48 kHz. */ - int16_t buffer16[kWebRtcOpusMaxFrameSize]; int decoded_samples; - int16_t output_samples; int i; - /* Decode to a temporary buffer. */ decoded_samples = DecodeNative(inst->decoder_right, encoded, encoded_bytes, - kWebRtcOpusMaxFrameSizePerChannel, buffer16, + kWebRtcOpusMaxFrameSizePerChannel, decoded, audio_type); if (decoded_samples < 0) { return -1; } if (inst->channels == 2) { /* The parameter |decoded_samples| holds the number of samples pairs, in - * case of stereo. Number of samples in |buffer16| equals |decoded_samples| + * case of stereo. Number of samples in |decoded| equals |decoded_samples| * times 2. */ for (i = 0; i < decoded_samples; i++) { /* Take every second sample, starting at the second sample. This gives * the right channel. */ - buffer16[i] = buffer16[i * 2 + 1]; + decoded[i] = decoded[i * 2 + 1]; } } else { /* Decode slave should never be called for mono packets. */ return -1; } - /* Resample from 48 kHz to 32 kHz. */ - output_samples = WebRtcOpus_Resample48to32(buffer16, decoded_samples, - inst->state_48_32_right, decoded); - return output_samples; + return decoded_samples; } int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded, int16_t number_of_lost_frames) { - int16_t buffer[kWebRtcOpusMaxFrameSize]; int16_t audio_type = 0; int decoded_samples; - int resampled_samples; int plc_samples; - /* If mono case, just do a regular call to the plc function, before - * resampling. - * If stereo, we need to de-interleave the stereo output into blocks with - * left and right channel. Each block is resampled to 32 kHz, and then - * interleaved again. */ - - /* Decode to a temporary buffer. The number of samples we ask for is - * |number_of_lost_frames| times |prev_decoded_samples_|. Limit the number - * of samples to maximum |kWebRtcOpusMaxFrameSizePerChannel|. */ + /* The number of samples we ask for is |number_of_lost_frames| times + * |prev_decoded_samples_|. Limit the number of samples to maximum + * |kWebRtcOpusMaxFrameSizePerChannel|. */ plc_samples = number_of_lost_frames * inst->prev_decoded_samples; plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ? plc_samples : kWebRtcOpusMaxFrameSizePerChannel; decoded_samples = DecodeNative(inst->decoder_left, NULL, 0, plc_samples, - buffer, &audio_type); + decoded, &audio_type); if (decoded_samples < 0) { return -1; } - if (inst->channels == 2) { - /* De-interleave and resample. */ - resampled_samples = WebRtcOpus_DeInterleaveResample(inst, - buffer, - decoded_samples, - decoded); - } else { - /* Resample from 48 kHz to 32 kHz. Filter state memory for left channel is - * used for mono signals. */ - resampled_samples = WebRtcOpus_Resample48to32(buffer, - decoded_samples, - inst->state_48_32_left, - decoded); - } - - return resampled_samples; + return decoded_samples; } int16_t WebRtcOpus_DecodePlcMaster(OpusDecInst* inst, int16_t* decoded, int16_t number_of_lost_frames) { - int16_t buffer[kWebRtcOpusMaxFrameSize]; int decoded_samples; - int resampled_samples; int16_t audio_type = 0; int plc_samples; int i; @@ -517,42 +364,35 @@ int16_t WebRtcOpus_DecodePlcMaster(OpusDecInst* inst, int16_t* decoded, * output. This is to make stereo work with the current setup of NetEQ, which * requires two calls to the decoder to produce stereo. */ - /* Decode to a temporary buffer. The number of samples we ask for is - * |number_of_lost_frames| times |prev_decoded_samples_|. Limit the number - * of samples to maximum |kWebRtcOpusMaxFrameSizePerChannel|. */ + /* The number of samples we ask for is |number_of_lost_frames| times + * |prev_decoded_samples_|. Limit the number of samples to maximum + * |kWebRtcOpusMaxFrameSizePerChannel|. */ plc_samples = number_of_lost_frames * inst->prev_decoded_samples; plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ? plc_samples : kWebRtcOpusMaxFrameSizePerChannel; decoded_samples = DecodeNative(inst->decoder_left, NULL, 0, plc_samples, - buffer, &audio_type); + decoded, &audio_type); if (decoded_samples < 0) { return -1; } if (inst->channels == 2) { /* The parameter |decoded_samples| holds the number of sample pairs, in - * case of stereo. The original number of samples in |buffer| equals + * case of stereo. The original number of samples in |decoded| equals * |decoded_samples| times 2. */ for (i = 0; i < decoded_samples; i++) { /* Take every second sample, starting at the first sample. This gives * the left channel. */ - buffer[i] = buffer[i * 2]; + decoded[i] = decoded[i * 2]; } } - /* Resample from 48 kHz to 32 kHz for left channel. */ - resampled_samples = WebRtcOpus_Resample48to32(buffer, - decoded_samples, - inst->state_48_32_left, - decoded); - return resampled_samples; + return decoded_samples; } int16_t WebRtcOpus_DecodePlcSlave(OpusDecInst* inst, int16_t* decoded, int16_t number_of_lost_frames) { - int16_t buffer[kWebRtcOpusMaxFrameSize]; int decoded_samples; - int resampled_samples; int16_t audio_type = 0; int plc_samples; int i; @@ -563,44 +403,35 @@ int16_t WebRtcOpus_DecodePlcSlave(OpusDecInst* inst, int16_t* decoded, return -1; } - /* Decode to a temporary buffer. The number of samples we ask for is - * |number_of_lost_frames| times |prev_decoded_samples_|. Limit the number - * of samples to maximum |kWebRtcOpusMaxFrameSizePerChannel|. */ + /* The number of samples we ask for is |number_of_lost_frames| times + * |prev_decoded_samples_|. Limit the number of samples to maximum + * |kWebRtcOpusMaxFrameSizePerChannel|. */ plc_samples = number_of_lost_frames * inst->prev_decoded_samples; plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ? plc_samples : kWebRtcOpusMaxFrameSizePerChannel; decoded_samples = DecodeNative(inst->decoder_right, NULL, 0, plc_samples, - buffer, &audio_type); + decoded, &audio_type); if (decoded_samples < 0) { return -1; } /* The parameter |decoded_samples| holds the number of sample pairs, - * The original number of samples in |buffer| equals |decoded_samples| + * The original number of samples in |decoded| equals |decoded_samples| * times 2. */ for (i = 0; i < decoded_samples; i++) { /* Take every second sample, starting at the second sample. This gives * the right channel. */ - buffer[i] = buffer[i * 2 + 1]; + decoded[i] = decoded[i * 2 + 1]; } - /* Resample from 48 kHz to 32 kHz for left channel. */ - resampled_samples = WebRtcOpus_Resample48to32(buffer, - decoded_samples, - inst->state_48_32_right, - decoded); - return resampled_samples; + return decoded_samples; } int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded, int16_t encoded_bytes, int16_t* decoded, int16_t* audio_type) { - /* |buffer| is big enough for 120 ms (the largest Opus packet size) of stereo - * audio at 48 kHz. */ - int16_t buffer[kWebRtcOpusMaxFrameSize]; int16_t* coded = (int16_t*)encoded; int decoded_samples; - int resampled_samples; int fec_samples; if (WebRtcOpus_PacketHasFec(encoded, encoded_bytes) != 1) { @@ -609,33 +440,13 @@ int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded, fec_samples = opus_packet_get_samples_per_frame(encoded, 48000); - /* Decode to a temporary buffer. */ decoded_samples = DecodeFec(inst->decoder_left, coded, encoded_bytes, - fec_samples, buffer, audio_type); + fec_samples, decoded, audio_type); if (decoded_samples < 0) { return -1; } - /* If mono case, just do a regular call to the decoder. - * If stereo, we need to de-interleave the stereo output into blocks with - * left and right channel. Each block is resampled to 32 kHz, and then - * interleaved again. */ - if (inst->channels == 2) { - /* De-interleave and resample. */ - resampled_samples = WebRtcOpus_DeInterleaveResample(inst, - buffer, - decoded_samples, - decoded); - } else { - /* Resample from 48 kHz to 32 kHz. Filter state memory for left channel is - * used for mono signals. */ - resampled_samples = WebRtcOpus_Resample48to32(buffer, - decoded_samples, - inst->state_48_32_left, - decoded); - } - - return resampled_samples; + return decoded_samples; } int WebRtcOpus_DurationEst(OpusDecInst* inst, @@ -652,10 +463,6 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst, /* Invalid payload duration. */ return 0; } - /* Compensate for the down-sampling from 48 kHz to 32 kHz. - * This should be removed when the resampling in WebRtcOpus_Decode is - * removed. */ - samples = samples * 2 / 3; return samples; } @@ -671,10 +478,6 @@ int WebRtcOpus_FecDurationEst(const uint8_t* payload, /* Invalid payload duration. */ return 0; } - /* Compensate for the down-sampling from 48 kHz to 32 kHz. - * This should be removed when the resampling in WebRtcOpus_Decode is - * removed. */ - samples = samples * 2 / 3; return samples; } diff --git a/modules/audio_coding/codecs/opus/opus_speed_test.cc b/modules/audio_coding/codecs/opus/opus_speed_test.cc index 16099c6d..e2439cf5 100644 --- a/modules/audio_coding/codecs/opus/opus_speed_test.cc +++ b/modules/audio_coding/codecs/opus/opus_speed_test.cc @@ -18,8 +18,7 @@ using ::testing::ValuesIn; namespace webrtc { static const int kOpusBlockDurationMs = 20; -static const int kOpusInputSamplingKhz = 48; -static const int kOpustOutputSamplingKhz = 32; +static const int kOpusSamplingKhz = 48; class OpusSpeedTest : public AudioCodecSpeedTest { protected: @@ -36,8 +35,8 @@ class OpusSpeedTest : public AudioCodecSpeedTest { OpusSpeedTest::OpusSpeedTest() : AudioCodecSpeedTest(kOpusBlockDurationMs, - kOpusInputSamplingKhz, - kOpustOutputSamplingKhz), + kOpusSamplingKhz, + kOpusSamplingKhz), opus_encoder_(NULL), opus_decoder_(NULL) { } diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc index ed876cd1..2ec77a53 100644 --- a/modules/audio_coding/codecs/opus/opus_unittest.cc +++ b/modules/audio_coding/codecs/opus/opus_unittest.cc @@ -19,9 +19,13 @@ struct WebRtcOpusDecInst; namespace webrtc { // Number of samples in a 60 ms stereo frame, sampled at 48 kHz. -const int kOpusNumberOfSamples = 480 * 6 * 2; +const int kOpusMaxFrameSamples = 48 * 60 * 2; // Maximum number of bytes in output bitstream. const size_t kMaxBytes = 1000; +// Number of samples-per-channel in a 20 ms frame, sampled at 48 kHz. +const int kOpus20msFrameSamples = 48 * 20; +// Number of samples-per-channel in a 10 ms frame, sampled at 48 kHz. +const int kOpus10msFrameSamples = 48 * 10; class OpusTest : public ::testing::Test { protected: @@ -35,8 +39,8 @@ class OpusTest : public ::testing::Test { WebRtcOpusDecInst* opus_stereo_decoder_; WebRtcOpusDecInst* opus_stereo_decoder_new_; - int16_t speech_data_[kOpusNumberOfSamples]; - int16_t output_data_[kOpusNumberOfSamples]; + int16_t speech_data_[kOpusMaxFrameSamples]; + int16_t output_data_[kOpusMaxFrameSamples]; uint8_t bitstream_[kMaxBytes]; }; @@ -50,17 +54,14 @@ OpusTest::OpusTest() } void OpusTest::SetUp() { - // Read some samples from a speech file, to be used in the encode test. - // In this test we do not care that the sampling frequency of the file is - // really 32000 Hz. We pretend that it is 48000 Hz. FILE* input_file; const std::string file_name = - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm"); input_file = fopen(file_name.c_str(), "rb"); ASSERT_TRUE(input_file != NULL); - ASSERT_EQ(kOpusNumberOfSamples, + ASSERT_EQ(kOpusMaxFrameSamples, static_cast<int32_t>(fread(speech_data_, sizeof(int16_t), - kOpusNumberOfSamples, input_file))); + kOpusMaxFrameSamples, input_file))); fclose(input_file); input_file = NULL; } @@ -114,21 +115,24 @@ TEST_F(OpusTest, OpusEncodeDecodeMono) { // Encode & decode. int16_t encoded_bytes; int16_t audio_type; - int16_t output_data_decode_new[kOpusNumberOfSamples]; - int16_t output_data_decode[kOpusNumberOfSamples]; + int16_t output_data_decode_new[kOpusMaxFrameSamples]; + int16_t output_data_decode[kOpusMaxFrameSamples]; int16_t* coded = reinterpret_cast<int16_t*>(bitstream_); - encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_, 960, - kMaxBytes, bitstream_); - EXPECT_EQ(640, WebRtcOpus_DecodeNew(opus_mono_decoder_new_, bitstream_, - encoded_bytes, output_data_decode_new, - &audio_type)); - EXPECT_EQ(640, WebRtcOpus_Decode(opus_mono_decoder_, coded, - encoded_bytes, output_data_decode, - &audio_type)); + encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_, + kOpus20msFrameSamples, kMaxBytes, + bitstream_); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodeNew(opus_mono_decoder_new_, bitstream_, + encoded_bytes, output_data_decode_new, + &audio_type)); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_Decode(opus_mono_decoder_, coded, + encoded_bytes, output_data_decode, + &audio_type)); // Data in |output_data_decode_new| should be the same as in // |output_data_decode|. - for (int i = 0; i < 640; i++) { + for (int i = 0; i < kOpus20msFrameSamples; i++) { EXPECT_EQ(output_data_decode_new[i], output_data_decode[i]); } @@ -154,26 +158,30 @@ TEST_F(OpusTest, OpusEncodeDecodeStereo) { // Encode & decode. int16_t encoded_bytes; int16_t audio_type; - int16_t output_data_decode_new[kOpusNumberOfSamples]; - int16_t output_data_decode[kOpusNumberOfSamples]; - int16_t output_data_decode_slave[kOpusNumberOfSamples]; + int16_t output_data_decode_new[kOpusMaxFrameSamples]; + int16_t output_data_decode[kOpusMaxFrameSamples]; + int16_t output_data_decode_slave[kOpusMaxFrameSamples]; int16_t* coded = reinterpret_cast<int16_t*>(bitstream_); - encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 960, - kMaxBytes, bitstream_); - EXPECT_EQ(640, WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_, - encoded_bytes, output_data_decode_new, - &audio_type)); - EXPECT_EQ(640, WebRtcOpus_Decode(opus_stereo_decoder_, coded, - encoded_bytes, output_data_decode, + encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, + kOpus20msFrameSamples, kMaxBytes, + bitstream_); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_, + encoded_bytes, output_data_decode_new, + &audio_type)); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_Decode(opus_stereo_decoder_, coded, + encoded_bytes, output_data_decode, + &audio_type)); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded, + encoded_bytes, output_data_decode_slave, &audio_type)); - EXPECT_EQ(640, WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded, - encoded_bytes, output_data_decode_slave, - &audio_type)); // Data in |output_data_decode_new| should be the same as in // |output_data_decode| and |output_data_decode_slave| interleaved to a // stereo signal. - for (int i = 0; i < 640; i++) { + for (int i = 0; i < kOpus20msFrameSamples; i++) { EXPECT_EQ(output_data_decode_new[i * 2], output_data_decode[i]); EXPECT_EQ(output_data_decode_new[i * 2 + 1], output_data_decode_slave[i]); } @@ -234,26 +242,30 @@ TEST_F(OpusTest, OpusDecodeInit) { // Encode & decode. int16_t encoded_bytes; int16_t audio_type; - int16_t output_data_decode_new[kOpusNumberOfSamples]; - int16_t output_data_decode[kOpusNumberOfSamples]; - int16_t output_data_decode_slave[kOpusNumberOfSamples]; + int16_t output_data_decode_new[kOpusMaxFrameSamples]; + int16_t output_data_decode[kOpusMaxFrameSamples]; + int16_t output_data_decode_slave[kOpusMaxFrameSamples]; int16_t* coded = reinterpret_cast<int16_t*>(bitstream_); - encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 960, - kMaxBytes, bitstream_); - EXPECT_EQ(640, WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_, - encoded_bytes, output_data_decode_new, - &audio_type)); - EXPECT_EQ(640, WebRtcOpus_Decode(opus_stereo_decoder_, coded, - encoded_bytes, output_data_decode, + encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, + kOpus20msFrameSamples, kMaxBytes, + bitstream_); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_, + encoded_bytes, output_data_decode_new, + &audio_type)); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_Decode(opus_stereo_decoder_, coded, + encoded_bytes, output_data_decode, + &audio_type)); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded, + encoded_bytes, output_data_decode_slave, &audio_type)); - EXPECT_EQ(640, WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded, - encoded_bytes, output_data_decode_slave, - &audio_type)); // Data in |output_data_decode_new| should be the same as in // |output_data_decode| and |output_data_decode_slave| interleaved to a // stereo signal. - for (int i = 0; i < 640; i++) { + for (int i = 0; i < kOpus20msFrameSamples; i++) { EXPECT_EQ(output_data_decode_new[i * 2], output_data_decode[i]); EXPECT_EQ(output_data_decode_new[i * 2 + 1], output_data_decode_slave[i]); } @@ -262,20 +274,23 @@ TEST_F(OpusTest, OpusDecodeInit) { EXPECT_EQ(0, WebRtcOpus_DecoderInit(opus_stereo_decoder_)); EXPECT_EQ(0, WebRtcOpus_DecoderInitSlave(opus_stereo_decoder_)); - EXPECT_EQ(640, WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_, - encoded_bytes, output_data_decode_new, - &audio_type)); - EXPECT_EQ(640, WebRtcOpus_Decode(opus_stereo_decoder_, coded, - encoded_bytes, output_data_decode, + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_, + encoded_bytes, output_data_decode_new, + &audio_type)); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_Decode(opus_stereo_decoder_, coded, + encoded_bytes, output_data_decode, + &audio_type)); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded, + encoded_bytes, output_data_decode_slave, &audio_type)); - EXPECT_EQ(640, WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded, - encoded_bytes, output_data_decode_slave, - &audio_type)); // Data in |output_data_decode_new| should be the same as in // |output_data_decode| and |output_data_decode_slave| interleaved to a // stereo signal. - for (int i = 0; i < 640; i++) { + for (int i = 0; i < kOpus20msFrameSamples; i++) { EXPECT_EQ(output_data_decode_new[i * 2], output_data_decode[i]); EXPECT_EQ(output_data_decode_new[i * 2 + 1], output_data_decode_slave[i]); } @@ -344,27 +359,31 @@ TEST_F(OpusTest, OpusDecodePlcMono) { // Encode & decode. int16_t encoded_bytes; int16_t audio_type; - int16_t output_data_decode_new[kOpusNumberOfSamples]; - int16_t output_data_decode[kOpusNumberOfSamples]; + int16_t output_data_decode_new[kOpusMaxFrameSamples]; + int16_t output_data_decode[kOpusMaxFrameSamples]; int16_t* coded = reinterpret_cast<int16_t*>(bitstream_); - encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_, 960, - kMaxBytes, bitstream_); - EXPECT_EQ(640, WebRtcOpus_DecodeNew(opus_mono_decoder_new_, bitstream_, - encoded_bytes, output_data_decode_new, - &audio_type)); - EXPECT_EQ(640, WebRtcOpus_Decode(opus_mono_decoder_, coded, - encoded_bytes, output_data_decode, - &audio_type)); + encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_, + kOpus20msFrameSamples, kMaxBytes, + bitstream_); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodeNew(opus_mono_decoder_new_, bitstream_, + encoded_bytes, output_data_decode_new, + &audio_type)); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_Decode(opus_mono_decoder_, coded, + encoded_bytes, output_data_decode, + &audio_type)); // Call decoder PLC for both versions of the decoder. - int16_t plc_buffer[kOpusNumberOfSamples]; - int16_t plc_buffer_new[kOpusNumberOfSamples]; - EXPECT_EQ(640, WebRtcOpus_DecodePlcMaster(opus_mono_decoder_, plc_buffer, 1)); - EXPECT_EQ(640, WebRtcOpus_DecodePlc(opus_mono_decoder_new_, - plc_buffer_new, 1)); + int16_t plc_buffer[kOpusMaxFrameSamples]; + int16_t plc_buffer_new[kOpusMaxFrameSamples]; + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodePlcMaster(opus_mono_decoder_, plc_buffer, 1)); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodePlc(opus_mono_decoder_new_, plc_buffer_new, 1)); // Data in |plc_buffer| should be the same as in |plc_buffer_new|. - for (int i = 0; i < 640; i++) { + for (int i = 0; i < kOpus20msFrameSamples; i++) { EXPECT_EQ(plc_buffer[i], plc_buffer_new[i]); } @@ -391,36 +410,42 @@ TEST_F(OpusTest, OpusDecodePlcStereo) { // Encode & decode. int16_t encoded_bytes; int16_t audio_type; - int16_t output_data_decode_new[kOpusNumberOfSamples]; - int16_t output_data_decode[kOpusNumberOfSamples]; - int16_t output_data_decode_slave[kOpusNumberOfSamples]; + int16_t output_data_decode_new[kOpusMaxFrameSamples]; + int16_t output_data_decode[kOpusMaxFrameSamples]; + int16_t output_data_decode_slave[kOpusMaxFrameSamples]; int16_t* coded = reinterpret_cast<int16_t*>(bitstream_); - encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 960, - kMaxBytes, bitstream_); - EXPECT_EQ(640, WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_, - encoded_bytes, output_data_decode_new, - &audio_type)); - EXPECT_EQ(640, WebRtcOpus_Decode(opus_stereo_decoder_, coded, - encoded_bytes, output_data_decode, + encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, + kOpus20msFrameSamples, kMaxBytes, + bitstream_); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_, + encoded_bytes, output_data_decode_new, + &audio_type)); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_Decode(opus_stereo_decoder_, coded, + encoded_bytes, output_data_decode, + &audio_type)); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded, + encoded_bytes, + output_data_decode_slave, &audio_type)); - EXPECT_EQ(640, WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded, - encoded_bytes, - output_data_decode_slave, - &audio_type)); // Call decoder PLC for both versions of the decoder. - int16_t plc_buffer_left[kOpusNumberOfSamples]; - int16_t plc_buffer_right[kOpusNumberOfSamples]; - int16_t plc_buffer_new[kOpusNumberOfSamples]; - EXPECT_EQ(640, WebRtcOpus_DecodePlcMaster(opus_stereo_decoder_, - plc_buffer_left, 1)); - EXPECT_EQ(640, WebRtcOpus_DecodePlcSlave(opus_stereo_decoder_, - plc_buffer_right, 1)); - EXPECT_EQ(640, WebRtcOpus_DecodePlc(opus_stereo_decoder_new_, plc_buffer_new, - 1)); + int16_t plc_buffer_left[kOpusMaxFrameSamples]; + int16_t plc_buffer_right[kOpusMaxFrameSamples]; + int16_t plc_buffer_new[kOpusMaxFrameSamples]; + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodePlcMaster(opus_stereo_decoder_, + plc_buffer_left, 1)); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodePlcSlave(opus_stereo_decoder_, + plc_buffer_right, 1)); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DecodePlc(opus_stereo_decoder_new_, plc_buffer_new, 1)); // Data in |plc_buffer_left| and |plc_buffer_right|should be the same as the // interleaved samples in |plc_buffer_new|. - for (int i = 0, j = 0; i < 640; i++) { + for (int i = 0, j = 0; i < kOpus20msFrameSamples; i++) { EXPECT_EQ(plc_buffer_left[i], plc_buffer_new[j++]); EXPECT_EQ(plc_buffer_right[i], plc_buffer_new[j++]); } @@ -437,21 +462,23 @@ TEST_F(OpusTest, OpusDurationEstimation) { EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2)); EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2)); - // Encode with different packet sizes (input 48 kHz, output in 32 kHz). int16_t encoded_bytes; // 10 ms. - encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 480, - kMaxBytes, bitstream_); - EXPECT_EQ(320, WebRtcOpus_DurationEst(opus_stereo_decoder_, bitstream_, - encoded_bytes)); + encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, + kOpus10msFrameSamples, kMaxBytes, + bitstream_); + EXPECT_EQ(kOpus10msFrameSamples, + WebRtcOpus_DurationEst(opus_stereo_decoder_, bitstream_, + encoded_bytes)); // 20 ms - encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 960, - kMaxBytes, bitstream_); - EXPECT_EQ(640, WebRtcOpus_DurationEst(opus_stereo_decoder_, bitstream_, - encoded_bytes)); - + encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, + kOpus20msFrameSamples, kMaxBytes, + bitstream_); + EXPECT_EQ(kOpus20msFrameSamples, + WebRtcOpus_DurationEst(opus_stereo_decoder_, bitstream_, + encoded_bytes)); // Free memory. EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_stereo_encoder_)); diff --git a/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/modules/audio_coding/main/acm2/audio_coding_module_impl.cc index f2410b7d..26f5b542 100644 --- a/modules/audio_coding/main/acm2/audio_coding_module_impl.cc +++ b/modules/audio_coding/main/acm2/audio_coding_module_impl.cc @@ -1618,14 +1618,8 @@ int AudioCodingModuleImpl::ReceiveFrequency() const { int codec_id = receiver_.last_audio_codec_id(); - int sample_rate_hz; - if (codec_id < 0) - sample_rate_hz = receiver_.current_sample_rate_hz(); - else - sample_rate_hz = ACMCodecDB::database_[codec_id].plfreq; - - // TODO(tlegrand): Remove this option when we have full 48 kHz support. - return (sample_rate_hz > 32000) ? 32000 : sample_rate_hz; + return codec_id < 0 ? receiver_.current_sample_rate_hz() : + ACMCodecDB::database_[codec_id].plfreq; } // Get current playout frequency. diff --git a/modules/audio_coding/main/test/opus_test.cc b/modules/audio_coding/main/test/opus_test.cc index 261eb613..398d59da 100644 --- a/modules/audio_coding/main/test/opus_test.cc +++ b/modules/audio_coding/main/test/opus_test.cc @@ -218,6 +218,8 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate, int written_samples = 0; int read_samples = 0; int decoded_samples = 0; + bool first_packet = true; + uint32_t start_time_stamp = 0; channel->reset_payload_size(); counter_ = 0; @@ -324,6 +326,10 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate, // Send data to the channel. "channel" will handle the loss simulation. channel->SendData(kAudioFrameSpeech, payload_type_, rtp_timestamp_, bitstream, bitstream_len_byte, NULL); + if (first_packet) { + first_packet = false; + start_time_stamp = rtp_timestamp_; + } rtp_timestamp_ += frame_length; read_samples += frame_length * channels; } @@ -344,9 +350,11 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate, // Write stand-alone speech to file. out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels); - // Number of channels should be the same for both stand-alone and - // ACM-decoding. - EXPECT_EQ(audio_frame.num_channels_, channels); + if (audio_frame.timestamp_ > start_time_stamp) { + // Number of channels should be the same for both stand-alone and + // ACM-decoding. + EXPECT_EQ(audio_frame.num_channels_, channels); + } decoded_samples = 0; } @@ -367,13 +375,13 @@ void OpusTest::OpenOutFile(int test_number) { file_stream << webrtc::test::OutputPath() << "opustest_out_" << test_number << ".pcm"; file_name = file_stream.str(); - out_file_.Open(file_name, 32000, "wb"); + out_file_.Open(file_name, 48000, "wb"); file_stream.str(""); file_name = file_stream.str(); file_stream << webrtc::test::OutputPath() << "opusstandalone_out_" << test_number << ".pcm"; file_name = file_stream.str(); - out_file_standalone_.Open(file_name, 32000, "wb"); + out_file_standalone_.Open(file_name, 48000, "wb"); } } // namespace webrtc diff --git a/modules/audio_coding/neteq/audio_decoder.cc b/modules/audio_coding/neteq/audio_decoder.cc index f539bb2e..0fdaa44b 100644 --- a/modules/audio_coding/neteq/audio_decoder.cc +++ b/modules/audio_coding/neteq/audio_decoder.cc @@ -162,7 +162,7 @@ int AudioDecoder::CodecSampleRateHz(NetEqDecoder codec_type) { #ifdef WEBRTC_CODEC_OPUS case kDecoderOpus: case kDecoderOpus_2ch: { - return 32000; + return 48000; } #endif case kDecoderCNGswb48kHz: { diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc index 05684ac7..687a733c 100644 --- a/modules/audio_coding/neteq/audio_decoder_unittest.cc +++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc @@ -602,7 +602,7 @@ class AudioDecoderCeltStereoTest : public AudioDecoderTest { class AudioDecoderOpusTest : public AudioDecoderTest { protected: AudioDecoderOpusTest() : AudioDecoderTest() { - frame_size_ = 320; + frame_size_ = 480; data_length_ = 10 * frame_size_; decoder_ = new AudioDecoderOpus(kDecoderOpus); assert(decoder_); @@ -613,75 +613,69 @@ class AudioDecoderOpusTest : public AudioDecoderTest { WebRtcOpus_EncoderFree(encoder_); } - virtual void InitEncoder() {} - - virtual int EncodeFrame(const int16_t* input, size_t input_len_samples, - uint8_t* output) { + virtual void SetUp() OVERRIDE { + AudioDecoderTest::SetUp(); // Upsample from 32 to 48 kHz. + // Because Opus is 48 kHz codec but the input file is 32 kHz, so the data + // read in |AudioDecoderTest::SetUp| has to be upsampled. + // |AudioDecoderTest::SetUp| has read |data_length_| samples, which is more + // than necessary after upsampling, so the end of audio that has been read + // is unused and the end of the buffer is overwritten by the resampled data. Resampler rs; rs.Reset(32000, 48000, kResamplerSynchronous); - const int max_resamp_len_samples = static_cast<int>(input_len_samples) * - 3 / 2; - int16_t* resamp_input = new int16_t[max_resamp_len_samples]; + const int before_resamp_len_samples = static_cast<int>(data_length_) * 2 + / 3; + int16_t* before_resamp_input = new int16_t[before_resamp_len_samples]; + memcpy(before_resamp_input, input_, + sizeof(int16_t) * before_resamp_len_samples); int resamp_len_samples; - EXPECT_EQ(0, rs.Push(input, static_cast<int>(input_len_samples), - resamp_input, max_resamp_len_samples, + EXPECT_EQ(0, rs.Push(before_resamp_input, before_resamp_len_samples, + input_, static_cast<int>(data_length_), resamp_len_samples)); - EXPECT_EQ(max_resamp_len_samples, resamp_len_samples); - int enc_len_bytes = - WebRtcOpus_Encode(encoder_, resamp_input, resamp_len_samples, - static_cast<int>(data_length_), output); + EXPECT_EQ(static_cast<int>(data_length_), resamp_len_samples); + delete[] before_resamp_input; + } + + virtual void InitEncoder() {} + + virtual int EncodeFrame(const int16_t* input, size_t input_len_samples, + uint8_t* output) OVERRIDE { + int enc_len_bytes = WebRtcOpus_Encode(encoder_, const_cast<int16_t*>(input), + static_cast<int16_t>(input_len_samples), + static_cast<int16_t>(data_length_), output); EXPECT_GT(enc_len_bytes, 0); - delete [] resamp_input; return enc_len_bytes; } OpusEncInst* encoder_; }; -class AudioDecoderOpusStereoTest : public AudioDecoderTest { +class AudioDecoderOpusStereoTest : public AudioDecoderOpusTest { protected: - AudioDecoderOpusStereoTest() : AudioDecoderTest() { + AudioDecoderOpusStereoTest() : AudioDecoderOpusTest() { channels_ = 2; - frame_size_ = 320; - data_length_ = 10 * frame_size_; + WebRtcOpus_EncoderFree(encoder_); + delete decoder_; decoder_ = new AudioDecoderOpus(kDecoderOpus_2ch); assert(decoder_); WebRtcOpus_EncoderCreate(&encoder_, 2); } - ~AudioDecoderOpusStereoTest() { - WebRtcOpus_EncoderFree(encoder_); - } - - virtual void InitEncoder() {} - virtual int EncodeFrame(const int16_t* input, size_t input_len_samples, - uint8_t* output) { + uint8_t* output) OVERRIDE { // Create stereo by duplicating each sample in |input|. const int input_stereo_samples = static_cast<int>(input_len_samples) * 2; int16_t* input_stereo = new int16_t[input_stereo_samples]; for (size_t i = 0; i < input_len_samples; i++) input_stereo[i * 2] = input_stereo[i * 2 + 1] = input[i]; - // Upsample from 32 to 48 kHz. - Resampler rs; - rs.Reset(32000, 48000, kResamplerSynchronousStereo); - const int max_resamp_len_samples = input_stereo_samples * 3 / 2; - int16_t* resamp_input = new int16_t[max_resamp_len_samples]; - int resamp_len_samples; - EXPECT_EQ(0, rs.Push(input_stereo, input_stereo_samples, resamp_input, - max_resamp_len_samples, resamp_len_samples)); - EXPECT_EQ(max_resamp_len_samples, resamp_len_samples); - int enc_len_bytes = - WebRtcOpus_Encode(encoder_, resamp_input, resamp_len_samples / 2, - static_cast<int16_t>(data_length_), output); + + int enc_len_bytes = WebRtcOpus_Encode( + encoder_, input_stereo, static_cast<int16_t>(input_len_samples), + static_cast<int16_t>(data_length_), output); EXPECT_GT(enc_len_bytes, 0); - delete [] resamp_input; - delete [] input_stereo; + delete[] input_stereo; return enc_len_bytes; } - - OpusEncInst* encoder_; }; TEST_F(AudioDecoderPcmUTest, EncodeDecode) { @@ -871,11 +865,11 @@ TEST(AudioDecoder, CodecSampleRateHz) { EXPECT_EQ(8000, AudioDecoder::CodecSampleRateHz(kDecoderCNGnb)); EXPECT_EQ(16000, AudioDecoder::CodecSampleRateHz(kDecoderCNGwb)); EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderCNGswb32kHz)); + EXPECT_EQ(48000, AudioDecoder::CodecSampleRateHz(kDecoderOpus)); + EXPECT_EQ(48000, AudioDecoder::CodecSampleRateHz(kDecoderOpus_2ch)); // TODO(tlegrand): Change 32000 to 48000 below once ACM has 48 kHz support. EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderCNGswb48kHz)); EXPECT_EQ(-1, AudioDecoder::CodecSampleRateHz(kDecoderArbitrary)); - EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderOpus)); - EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderOpus_2ch)); #ifdef WEBRTC_CODEC_CELT EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderCELT_32)); EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderCELT_32_2ch)); diff --git a/modules/audio_coding/neteq/payload_splitter_unittest.cc b/modules/audio_coding/neteq/payload_splitter_unittest.cc index 5cde1bda..9d0aaa1d 100644 --- a/modules/audio_coding/neteq/payload_splitter_unittest.cc +++ b/modules/audio_coding/neteq/payload_splitter_unittest.cc @@ -743,7 +743,7 @@ TEST(FecPayloadSplitter, MixedPayload) { // Check first packet. packet = packet_list.front(); EXPECT_EQ(0, packet->header.payloadType); - EXPECT_EQ(kBaseTimestamp - 20 * 32, packet->header.timestamp); + EXPECT_EQ(kBaseTimestamp - 20 * 48, packet->header.timestamp); EXPECT_EQ(10, packet->payload_length); EXPECT_FALSE(packet->primary); delete [] packet->payload; diff --git a/modules/audio_coding/neteq/test/neteq_opus_fec_quality_test.cc b/modules/audio_coding/neteq/test/neteq_opus_fec_quality_test.cc index e8fd06a4..dee99b87 100644 --- a/modules/audio_coding/neteq/test/neteq_opus_fec_quality_test.cc +++ b/modules/audio_coding/neteq/test/neteq_opus_fec_quality_test.cc @@ -21,8 +21,7 @@ namespace webrtc { namespace test { static const int kOpusBlockDurationMs = 20; -static const int kOpusInputSamplingKhz = 48; -static const int kOpusOutputSamplingKhz = 32; +static const int kOpusSamplingKhz = 48; // Define switch for input file name. static bool ValidateInFilename(const char* flagname, const string& value) { @@ -128,8 +127,8 @@ class NetEqOpusFecQualityTest : public NetEqQualityTest { }; NetEqOpusFecQualityTest::NetEqOpusFecQualityTest() - : NetEqQualityTest(kOpusBlockDurationMs, kOpusInputSamplingKhz, - kOpusOutputSamplingKhz, + : NetEqQualityTest(kOpusBlockDurationMs, kOpusSamplingKhz, + kOpusSamplingKhz, (FLAGS_channels == 1) ? kDecoderOpus : kDecoderOpus_2ch, FLAGS_channels, FLAGS_in_filename, diff --git a/modules/audio_coding/neteq/timestamp_scaler.cc b/modules/audio_coding/neteq/timestamp_scaler.cc index 01890136..1809324b 100644 --- a/modules/audio_coding/neteq/timestamp_scaler.cc +++ b/modules/audio_coding/neteq/timestamp_scaler.cc @@ -48,8 +48,6 @@ uint32_t TimestampScaler::ToInternal(uint32_t external_timestamp, denominator_ = 1; break; } - case kDecoderOpus: - case kDecoderOpus_2ch: case kDecoderISACfb: case kDecoderCNGswb48kHz: { // Use timestamp scaling with factor 2/3 (32 kHz sample rate, but RTP diff --git a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc index 8cbbfa39..1cbbf7f3 100644 --- a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc +++ b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc @@ -252,10 +252,14 @@ TEST(TimestampScaler, TestG722Reset) { EXPECT_CALL(db, Die()); // Called when database object is deleted. } +// TODO(minyue): This test becomes trivial since Opus does not need a timestamp +// scaler. Therefore, this test may be removed in future. There is no harm to +// keep it, since it can be taken as a test case for the situation of a trivial +// timestamp scaler. TEST(TimestampScaler, TestOpusLargeStep) { MockDecoderDatabase db; DecoderDatabase::DecoderInfo info; - info.codec_type = kDecoderOpus; // Uses a factor 2/3 scaling. + info.codec_type = kDecoderOpus; static const uint8_t kRtpPayloadType = 17; EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType)) .WillRepeatedly(Return(&info)); @@ -273,8 +277,7 @@ TEST(TimestampScaler, TestOpusLargeStep) { scaler.ToInternal(external_timestamp, kRtpPayloadType)); // Scale back. EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp)); - // Internal timestamp should be incremented with twice the step. - internal_timestamp += 2 * kStep / 3; + internal_timestamp += kStep; } EXPECT_CALL(db, Die()); // Called when database object is deleted. @@ -283,7 +286,7 @@ TEST(TimestampScaler, TestOpusLargeStep) { TEST(TimestampScaler, TestIsacFbLargeStep) { MockDecoderDatabase db; DecoderDatabase::DecoderInfo info; - info.codec_type = kDecoderISACfb; // Uses a factor 2/3 scaling. + info.codec_type = kDecoderISACfb; static const uint8_t kRtpPayloadType = 17; EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType)) .WillRepeatedly(Return(&info)); @@ -301,7 +304,7 @@ TEST(TimestampScaler, TestIsacFbLargeStep) { scaler.ToInternal(external_timestamp, kRtpPayloadType)); // Scale back. EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp)); - // Internal timestamp should be incremented with twice the step. + // Internal timestamp should be incremented with two-thirds the step. internal_timestamp += 2 * kStep / 3; } diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/modules/audio_coding/neteq/tools/neteq_quality_test.cc index a80b1f88..83288557 100644 --- a/modules/audio_coding/neteq/tools/neteq_quality_test.cc +++ b/modules/audio_coding/neteq/tools/neteq_quality_test.cc @@ -158,7 +158,7 @@ bool NoLoss::Lost() { return false; } -UniformLoss::UniformLoss(int loss_rate) +UniformLoss::UniformLoss(double loss_rate) : loss_rate_(loss_rate) { } @@ -204,8 +204,6 @@ void NetEqQualityTest::SetUp() { // a full packet duration is drawn with a loss, |unit_loss_rate| fulfills // (1 - unit_loss_rate) ^ (block_duration_ms_ / kPacketLossTimeUnitMs) == // 1 - packet_loss_rate. - // |unit_loss_rate| is usually small. To increase its resolution, we - // magnify it by |RAND_MAX|. double unit_loss_rate = (1.0f - pow(1.0f - 0.01f * packet_loss_rate_, 1.0f / units)); loss_model_.reset(new UniformLoss(unit_loss_rate)); diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.h b/modules/audio_coding/neteq/tools/neteq_quality_test.h index 75d19ae6..e0a43b6f 100644 --- a/modules/audio_coding/neteq/tools/neteq_quality_test.h +++ b/modules/audio_coding/neteq/tools/neteq_quality_test.h @@ -38,9 +38,10 @@ class NoLoss : public LossModel { class UniformLoss : public LossModel { public: - UniformLoss(int loss_rate); + UniformLoss(double loss_rate); virtual bool Lost() OVERRIDE; void set_loss_rate(double loss_rate) { loss_rate_ = loss_rate; } + private: double loss_rate_; }; @@ -49,6 +50,7 @@ class GilbertElliotLoss : public LossModel { public: GilbertElliotLoss(double prob_trans_11, double prob_trans_01); virtual bool Lost() OVERRIDE; + private: // Prob. of losing current packet, when previous packet is lost. double prob_trans_11_; diff --git a/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc index 26ef3e88..6ef6166b 100644 --- a/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc +++ b/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc @@ -32,10 +32,13 @@ typedef std::list<ParticipantFramePair*> ParticipantFramePairList; // stereo at most. // // TODO(andrew): consider not modifying |frame| here. -void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame) { +void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) { assert(mixed_frame->num_channels_ >= frame->num_channels_); - // Divide by two to avoid saturation in the mixing. - *frame >>= 1; + if (use_limiter) { + // Divide by two to avoid saturation in the mixing. + // This is only meaningful if the limiter will be used. + *frame >>= 1; + } if (mixed_frame->num_channels_ > frame->num_channels_) { // We only support mono-to-stereo. assert(mixed_frame->num_channels_ == 2 && @@ -131,6 +134,7 @@ AudioConferenceMixerImpl::AudioConferenceMixerImpl(int id) _participantList(), _additionalParticipantList(), _numMixedParticipants(0), + use_limiter_(true), _timeStamp(0), _timeScheduler(kProcessPeriodicityInMs), _mixedAudioLevel(), @@ -308,6 +312,11 @@ int32_t AudioConferenceMixerImpl::Process() { _timeStamp += _sampleSize; + // We only use the limiter if it supports the output sample rate and + // we're actually mixing multiple streams. + use_limiter_ = _numMixedParticipants > 1 && + _outputFrequency <= kAudioProcMaxNativeSampleRateHz; + MixFromList(*mixedAudio, &mixList); MixAnonomouslyFromList(*mixedAudio, &additionalFramesList); MixAnonomouslyFromList(*mixedAudio, &rampOutList); @@ -946,16 +955,8 @@ int32_t AudioConferenceMixerImpl::MixFromList( if(audioFrameList->empty()) return 0; uint32_t position = 0; - if(_numMixedParticipants == 1) { - // No mixing required here; skip the saturation protection. - AudioFrame* audioFrame = audioFrameList->front(); - mixedAudio.CopyFrom(*audioFrame); - SetParticipantStatistics(&_scratchMixedParticipants[position], - *audioFrame); - return 0; - } - if (audioFrameList->size() == 1) { + if (_numMixedParticipants == 1) { mixedAudio.timestamp_ = audioFrameList->front()->timestamp_; mixedAudio.elapsed_time_ms_ = audioFrameList->front()->elapsed_time_ms_; } else { @@ -979,7 +980,7 @@ int32_t AudioConferenceMixerImpl::MixFromList( assert(false); position = 0; } - MixFrames(&mixedAudio, (*iter)); + MixFrames(&mixedAudio, (*iter), use_limiter_); SetParticipantStatistics(&_scratchMixedParticipants[position], **iter); @@ -999,24 +1000,17 @@ int32_t AudioConferenceMixerImpl::MixAnonomouslyFromList( if(audioFrameList->empty()) return 0; - if(_numMixedParticipants == 1) { - // No mixing required here; skip the saturation protection. - AudioFrame* audioFrame = audioFrameList->front(); - mixedAudio.CopyFrom(*audioFrame); - return 0; - } - for (AudioFrameList::const_iterator iter = audioFrameList->begin(); iter != audioFrameList->end(); ++iter) { - MixFrames(&mixedAudio, *iter); + MixFrames(&mixedAudio, *iter, use_limiter_); } return 0; } bool AudioConferenceMixerImpl::LimitMixedAudio(AudioFrame& mixedAudio) { - if(_numMixedParticipants == 1) { - return true; + if (!use_limiter_) { + return true; } // Smoothly limit the mixed frame. diff --git a/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h b/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h index 31dc71e5..44f4ff04 100644 --- a/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h +++ b/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h @@ -192,6 +192,9 @@ private: MixerParticipantList _additionalParticipantList; size_t _numMixedParticipants; + // Determines if we will use a limiter for clipping protection during + // mixing. + bool use_limiter_; uint32_t _timeStamp; diff --git a/modules/audio_device/audio_device.gypi b/modules/audio_device/audio_device.gypi index a64856b5..23f417f9 100644 --- a/modules/audio_device/audio_device.gypi +++ b/modules/audio_device/audio_device.gypi @@ -98,7 +98,7 @@ 'linux/audio_mixer_manager_alsa_linux.h', 'linux/latebindingsymboltable_linux.cc', 'linux/latebindingsymboltable_linux.h', - 'ios/audio_device_ios.cc', + 'ios/audio_device_ios.mm', 'ios/audio_device_ios.h', 'ios/audio_device_utility_ios.cc', 'ios/audio_device_utility_ios.h', @@ -175,7 +175,7 @@ }], ], }], - ['OS=="mac" or OS=="ios"', { + ['OS=="mac"', { 'link_settings': { 'libraries': [ '$(SDKROOT)/System/Library/Frameworks/AudioToolbox.framework', @@ -183,6 +183,19 @@ ], }, }], + ['OS=="ios"', { + 'xcode_settings': { + 'CLANG_ENABLE_OBJC_ARC': 'YES', + }, + 'link_settings': { + 'xcode_settings': { + 'OTHER_LDFLAGS': [ + '-framework AudioToolbox', + '-framework AVFoundation', + ], + }, + }, + }], ['OS=="win"', { 'link_settings': { 'libraries': [ diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc index 58411e3b..a8145603 100644 --- a/modules/audio_device/audio_device_impl.cc +++ b/modules/audio_device/audio_device_impl.cc @@ -349,15 +349,15 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() #if defined(WEBRTC_IOS) if (audioLayer == kPlatformDefaultAudio) { - // Create *iPhone Audio* implementation - ptrAudioDevice = new AudioDeviceIPhone(Id()); + // Create iOS Audio Device implementation. + ptrAudioDevice = new AudioDeviceIOS(Id()); WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "iPhone Audio APIs will be utilized"); } if (ptrAudioDevice != NULL) { - // Create the Mac implementation of the Device Utility. - ptrAudioDeviceUtility = new AudioDeviceUtilityIPhone(Id()); + // Create iOS Device Utility implementation. + ptrAudioDeviceUtility = new AudioDeviceUtilityIOS(Id()); } // END #if defined(WEBRTC_IOS) diff --git a/modules/audio_device/audio_device_utility.h b/modules/audio_device/audio_device_utility.h index eb3e623e..ebe06d1f 100644 --- a/modules/audio_device/audio_device_utility.h +++ b/modules/audio_device/audio_device_utility.h @@ -18,15 +18,15 @@ namespace webrtc class AudioDeviceUtility { -public: - static uint32_t GetTimeInMS(); - static void WaitForKey(); - static bool StringCompare(const char* str1, - const char* str2, - const uint32_t length); - virtual int32_t Init() = 0; + public: + static uint32_t GetTimeInMS(); + static void WaitForKey(); + static bool StringCompare(const char* str1, + const char* str2, + const uint32_t length); + virtual int32_t Init() = 0; - virtual ~AudioDeviceUtility() {} + virtual ~AudioDeviceUtility() {} }; } // namespace webrtc diff --git a/modules/audio_device/ios/audio_device_ios.h b/modules/audio_device/ios/audio_device_ios.h index 011b6acf..2a48845e 100644 --- a/modules/audio_device/ios/audio_device_ios.h +++ b/modules/audio_device/ios/audio_device_ios.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IPHONE_H -#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IPHONE_H +#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IOS_H +#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IOS_H #include <AudioUnit/AudioUnit.h> @@ -32,244 +32,241 @@ const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC / 100); // Number of 10 ms recording blocks in recording buffer const uint16_t N_REC_BUFFERS = 20; -class AudioDeviceIPhone : public AudioDeviceGeneric { -public: - AudioDeviceIPhone(const int32_t id); - ~AudioDeviceIPhone(); - - // Retrieve the currently utilized audio layer - virtual int32_t - ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const; - - // Main initializaton and termination - virtual int32_t Init(); - virtual int32_t Terminate(); - virtual bool Initialized() const; - - // Device enumeration - virtual int16_t PlayoutDevices(); - virtual int16_t RecordingDevices(); - virtual int32_t PlayoutDeviceName(uint16_t index, +class AudioDeviceIOS : public AudioDeviceGeneric { + public: + AudioDeviceIOS(const int32_t id); + ~AudioDeviceIOS(); + + // Retrieve the currently utilized audio layer + virtual int32_t ActiveAudioLayer( + AudioDeviceModule::AudioLayer& audioLayer) const; + + // Main initializaton and termination + virtual int32_t Init(); + virtual int32_t Terminate(); + virtual bool Initialized() const; + + // Device enumeration + virtual int16_t PlayoutDevices(); + virtual int16_t RecordingDevices(); + virtual int32_t PlayoutDeviceName(uint16_t index, + char name[kAdmMaxDeviceNameSize], + char guid[kAdmMaxGuidSize]); + virtual int32_t RecordingDeviceName(uint16_t index, char name[kAdmMaxDeviceNameSize], char guid[kAdmMaxGuidSize]); - virtual int32_t RecordingDeviceName(uint16_t index, - char name[kAdmMaxDeviceNameSize], - char guid[kAdmMaxGuidSize]); - - // Device selection - virtual int32_t SetPlayoutDevice(uint16_t index); - virtual int32_t - SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device); - virtual int32_t SetRecordingDevice(uint16_t index); - virtual int32_t SetRecordingDevice( - AudioDeviceModule::WindowsDeviceType device); - - // Audio transport initialization - virtual int32_t PlayoutIsAvailable(bool& available); - virtual int32_t InitPlayout(); - virtual bool PlayoutIsInitialized() const; - virtual int32_t RecordingIsAvailable(bool& available); - virtual int32_t InitRecording(); - virtual bool RecordingIsInitialized() const; - - // Audio transport control - virtual int32_t StartPlayout(); - virtual int32_t StopPlayout(); - virtual bool Playing() const; - virtual int32_t StartRecording(); - virtual int32_t StopRecording(); - virtual bool Recording() const; - - // Microphone Automatic Gain Control (AGC) - virtual int32_t SetAGC(bool enable); - virtual bool AGC() const; - - // Volume control based on the Windows Wave API (Windows only) - virtual int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight); - virtual int32_t WaveOutVolume(uint16_t& volumeLeft, - uint16_t& volumeRight) const; - - // Audio mixer initialization - virtual int32_t InitSpeaker(); - virtual bool SpeakerIsInitialized() const; - virtual int32_t InitMicrophone(); - virtual bool MicrophoneIsInitialized() const; - - // Speaker volume controls - virtual int32_t SpeakerVolumeIsAvailable(bool& available); - virtual int32_t SetSpeakerVolume(uint32_t volume); - virtual int32_t SpeakerVolume(uint32_t& volume) const; - virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const; - virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const; - virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const; - - // Microphone volume controls - virtual int32_t MicrophoneVolumeIsAvailable(bool& available); - virtual int32_t SetMicrophoneVolume(uint32_t volume); - virtual int32_t MicrophoneVolume(uint32_t& volume) const; - virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const; - virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const; - virtual int32_t - MicrophoneVolumeStepSize(uint16_t& stepSize) const; - - // Microphone mute control - virtual int32_t MicrophoneMuteIsAvailable(bool& available); - virtual int32_t SetMicrophoneMute(bool enable); - virtual int32_t MicrophoneMute(bool& enabled) const; - - // Speaker mute control - virtual int32_t SpeakerMuteIsAvailable(bool& available); - virtual int32_t SetSpeakerMute(bool enable); - virtual int32_t SpeakerMute(bool& enabled) const; - - // Microphone boost control - virtual int32_t MicrophoneBoostIsAvailable(bool& available); - virtual int32_t SetMicrophoneBoost(bool enable); - virtual int32_t MicrophoneBoost(bool& enabled) const; - - // Stereo support - virtual int32_t StereoPlayoutIsAvailable(bool& available); - virtual int32_t SetStereoPlayout(bool enable); - virtual int32_t StereoPlayout(bool& enabled) const; - virtual int32_t StereoRecordingIsAvailable(bool& available); - virtual int32_t SetStereoRecording(bool enable); - virtual int32_t StereoRecording(bool& enabled) const; - - // Delay information and control - virtual int32_t - SetPlayoutBuffer(const AudioDeviceModule::BufferType type, - uint16_t sizeMS); - virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type, - uint16_t& sizeMS) const; - virtual int32_t PlayoutDelay(uint16_t& delayMS) const; - virtual int32_t RecordingDelay(uint16_t& delayMS) const; - - // CPU load - virtual int32_t CPULoad(uint16_t& load) const; - -public: - virtual bool PlayoutWarning() const; - virtual bool PlayoutError() const; - virtual bool RecordingWarning() const; - virtual bool RecordingError() const; - virtual void ClearPlayoutWarning(); - virtual void ClearPlayoutError(); - virtual void ClearRecordingWarning(); - virtual void ClearRecordingError(); - -public: - virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer); - - // Reset Audio Deivce (for mobile devices only) - virtual int32_t ResetAudioDevice(); - - // enable or disable loud speaker (for iphone only) - virtual int32_t SetLoudspeakerStatus(bool enable); - virtual int32_t GetLoudspeakerStatus(bool& enabled) const; - -private: - void Lock() { - _critSect.Enter(); - } - - void UnLock() { - _critSect.Leave(); - } - - int32_t Id() { - return _id; - } - - // Init and shutdown - int32_t InitPlayOrRecord(); - int32_t ShutdownPlayOrRecord(); - - void UpdateRecordingDelay(); - void UpdatePlayoutDelay(); - - static OSStatus RecordProcess(void *inRefCon, - AudioUnitRenderActionFlags *ioActionFlags, - const AudioTimeStamp *timeStamp, - UInt32 inBusNumber, - UInt32 inNumberFrames, - AudioBufferList *ioData); - - static OSStatus PlayoutProcess(void *inRefCon, - AudioUnitRenderActionFlags *ioActionFlags, - const AudioTimeStamp *timeStamp, - UInt32 inBusNumber, - UInt32 inNumberFrames, - AudioBufferList *ioData); - - OSStatus RecordProcessImpl(AudioUnitRenderActionFlags *ioActionFlags, - const AudioTimeStamp *timeStamp, - uint32_t inBusNumber, - uint32_t inNumberFrames); - - OSStatus PlayoutProcessImpl(uint32_t inNumberFrames, + + // Device selection + virtual int32_t SetPlayoutDevice(uint16_t index); + virtual int32_t SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device); + virtual int32_t SetRecordingDevice(uint16_t index); + virtual int32_t SetRecordingDevice( + AudioDeviceModule::WindowsDeviceType device); + + // Audio transport initialization + virtual int32_t PlayoutIsAvailable(bool& available); + virtual int32_t InitPlayout(); + virtual bool PlayoutIsInitialized() const; + virtual int32_t RecordingIsAvailable(bool& available); + virtual int32_t InitRecording(); + virtual bool RecordingIsInitialized() const; + + // Audio transport control + virtual int32_t StartPlayout(); + virtual int32_t StopPlayout(); + virtual bool Playing() const; + virtual int32_t StartRecording(); + virtual int32_t StopRecording(); + virtual bool Recording() const; + + // Microphone Automatic Gain Control (AGC) + virtual int32_t SetAGC(bool enable); + virtual bool AGC() const; + + // Volume control based on the Windows Wave API (Windows only) + virtual int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight); + virtual int32_t WaveOutVolume(uint16_t& volumeLeft, + uint16_t& volumeRight) const; + + // Audio mixer initialization + virtual int32_t InitSpeaker(); + virtual bool SpeakerIsInitialized() const; + virtual int32_t InitMicrophone(); + virtual bool MicrophoneIsInitialized() const; + + // Speaker volume controls + virtual int32_t SpeakerVolumeIsAvailable(bool& available); + virtual int32_t SetSpeakerVolume(uint32_t volume); + virtual int32_t SpeakerVolume(uint32_t& volume) const; + virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const; + virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const; + virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const; + + // Microphone volume controls + virtual int32_t MicrophoneVolumeIsAvailable(bool& available); + virtual int32_t SetMicrophoneVolume(uint32_t volume); + virtual int32_t MicrophoneVolume(uint32_t& volume) const; + virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const; + virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const; + virtual int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const; + + // Microphone mute control + virtual int32_t MicrophoneMuteIsAvailable(bool& available); + virtual int32_t SetMicrophoneMute(bool enable); + virtual int32_t MicrophoneMute(bool& enabled) const; + + // Speaker mute control + virtual int32_t SpeakerMuteIsAvailable(bool& available); + virtual int32_t SetSpeakerMute(bool enable); + virtual int32_t SpeakerMute(bool& enabled) const; + + // Microphone boost control + virtual int32_t MicrophoneBoostIsAvailable(bool& available); + virtual int32_t SetMicrophoneBoost(bool enable); + virtual int32_t MicrophoneBoost(bool& enabled) const; + + // Stereo support + virtual int32_t StereoPlayoutIsAvailable(bool& available); + virtual int32_t SetStereoPlayout(bool enable); + virtual int32_t StereoPlayout(bool& enabled) const; + virtual int32_t StereoRecordingIsAvailable(bool& available); + virtual int32_t SetStereoRecording(bool enable); + virtual int32_t StereoRecording(bool& enabled) const; + + // Delay information and control + virtual int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type, + uint16_t sizeMS); + virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type, + uint16_t& sizeMS) const; + virtual int32_t PlayoutDelay(uint16_t& delayMS) const; + virtual int32_t RecordingDelay(uint16_t& delayMS) const; + + // CPU load + virtual int32_t CPULoad(uint16_t& load) const; + + public: + virtual bool PlayoutWarning() const; + virtual bool PlayoutError() const; + virtual bool RecordingWarning() const; + virtual bool RecordingError() const; + virtual void ClearPlayoutWarning(); + virtual void ClearPlayoutError(); + virtual void ClearRecordingWarning(); + virtual void ClearRecordingError(); + + public: + virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer); + + // Reset Audio Device (for mobile devices only) + virtual int32_t ResetAudioDevice(); + + // enable or disable loud speaker (for iphone only) + virtual int32_t SetLoudspeakerStatus(bool enable); + virtual int32_t GetLoudspeakerStatus(bool& enabled) const; + + private: + void Lock() { + _critSect.Enter(); + } + + void UnLock() { + _critSect.Leave(); + } + + int32_t Id() { + return _id; + } + + // Init and shutdown + int32_t InitPlayOrRecord(); + int32_t ShutdownPlayOrRecord(); + + void UpdateRecordingDelay(); + void UpdatePlayoutDelay(); + + static OSStatus RecordProcess(void *inRefCon, + AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *timeStamp, + UInt32 inBusNumber, + UInt32 inNumberFrames, AudioBufferList *ioData); - static bool RunCapture(void* ptrThis); - bool CaptureWorkerThread(); + static OSStatus PlayoutProcess(void *inRefCon, + AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *timeStamp, + UInt32 inBusNumber, + UInt32 inNumberFrames, + AudioBufferList *ioData); + + OSStatus RecordProcessImpl(AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *timeStamp, + uint32_t inBusNumber, + uint32_t inNumberFrames); + + OSStatus PlayoutProcessImpl(uint32_t inNumberFrames, + AudioBufferList *ioData); + + static bool RunCapture(void* ptrThis); + bool CaptureWorkerThread(); -private: - AudioDeviceBuffer* _ptrAudioBuffer; + private: + AudioDeviceBuffer* _ptrAudioBuffer; - CriticalSectionWrapper& _critSect; + CriticalSectionWrapper& _critSect; - ThreadWrapper* _captureWorkerThread; - uint32_t _captureWorkerThreadId; + ThreadWrapper* _captureWorkerThread; + uint32_t _captureWorkerThreadId; - int32_t _id; + int32_t _id; - AudioUnit _auVoiceProcessing; + AudioUnit _auVoiceProcessing; + void* _audioInterruptionObserver; -private: - bool _initialized; - bool _isShutDown; - bool _recording; - bool _playing; - bool _recIsInitialized; - bool _playIsInitialized; + private: + bool _initialized; + bool _isShutDown; + bool _recording; + bool _playing; + bool _recIsInitialized; + bool _playIsInitialized; - bool _recordingDeviceIsSpecified; - bool _playoutDeviceIsSpecified; - bool _micIsInitialized; - bool _speakerIsInitialized; + bool _recordingDeviceIsSpecified; + bool _playoutDeviceIsSpecified; + bool _micIsInitialized; + bool _speakerIsInitialized; - bool _AGC; + bool _AGC; - // The sampling rate to use with Audio Device Buffer - uint32_t _adbSampFreq; + // The sampling rate to use with Audio Device Buffer + uint32_t _adbSampFreq; - // Delay calculation - uint32_t _recordingDelay; - uint32_t _playoutDelay; - uint32_t _playoutDelayMeasurementCounter; - uint32_t _recordingDelayHWAndOS; - uint32_t _recordingDelayMeasurementCounter; + // Delay calculation + uint32_t _recordingDelay; + uint32_t _playoutDelay; + uint32_t _playoutDelayMeasurementCounter; + uint32_t _recordingDelayHWAndOS; + uint32_t _recordingDelayMeasurementCounter; - // Errors and warnings count - uint16_t _playWarning; - uint16_t _playError; - uint16_t _recWarning; - uint16_t _recError; + // Errors and warnings count + uint16_t _playWarning; + uint16_t _playError; + uint16_t _recWarning; + uint16_t _recError; - // Playout buffer, needed for 44.0 / 44.1 kHz mismatch - int16_t _playoutBuffer[ENGINE_PLAY_BUF_SIZE_IN_SAMPLES]; - uint32_t _playoutBufferUsed; // How much is filled + // Playout buffer, needed for 44.0 / 44.1 kHz mismatch + int16_t _playoutBuffer[ENGINE_PLAY_BUF_SIZE_IN_SAMPLES]; + uint32_t _playoutBufferUsed; // How much is filled - // Recording buffers - int16_t - _recordingBuffer[N_REC_BUFFERS][ENGINE_REC_BUF_SIZE_IN_SAMPLES]; - uint32_t _recordingLength[N_REC_BUFFERS]; - uint32_t _recordingSeqNumber[N_REC_BUFFERS]; - uint32_t _recordingCurrentSeq; + // Recording buffers + int16_t _recordingBuffer[N_REC_BUFFERS][ENGINE_REC_BUF_SIZE_IN_SAMPLES]; + uint32_t _recordingLength[N_REC_BUFFERS]; + uint32_t _recordingSeqNumber[N_REC_BUFFERS]; + uint32_t _recordingCurrentSeq; - // Current total size all data in buffers, used for delay estimate - uint32_t _recordingBufferTotalSize; + // Current total size all data in buffers, used for delay estimate + uint32_t _recordingBufferTotalSize; }; } // namespace webrtc -#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_IPHONE_H_ +#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IOS_H diff --git a/modules/audio_device/ios/audio_device_ios.cc b/modules/audio_device/ios/audio_device_ios.mm index 7a7189a2..19dcfdfb 100644 --- a/modules/audio_device/ios/audio_device_ios.cc +++ b/modules/audio_device/ios/audio_device_ios.mm @@ -8,7 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include <AudioToolbox/AudioServices.h> // AudioSession +#import <AVFoundation/AVFoundation.h> +#import <Foundation/Foundation.h> #include "webrtc/modules/audio_device/ios/audio_device_ios.h" @@ -16,7 +17,7 @@ #include "webrtc/system_wrappers/interface/trace.h" namespace webrtc { -AudioDeviceIPhone::AudioDeviceIPhone(const int32_t id) +AudioDeviceIOS::AudioDeviceIOS(const int32_t id) : _ptrAudioBuffer(NULL), _critSect(*CriticalSectionWrapper::CreateCriticalSection()), @@ -24,6 +25,7 @@ AudioDeviceIPhone::AudioDeviceIPhone(const int32_t id) _captureWorkerThreadId(0), _id(id), _auVoiceProcessing(NULL), + _audioInterruptionObserver(NULL), _initialized(false), _isShutDown(false), _recording(false), @@ -57,7 +59,7 @@ AudioDeviceIPhone::AudioDeviceIPhone(const int32_t id) memset(_recordingSeqNumber, 0, sizeof(_recordingSeqNumber)); } -AudioDeviceIPhone::~AudioDeviceIPhone() { +AudioDeviceIOS::~AudioDeviceIOS() { WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__); @@ -71,7 +73,7 @@ AudioDeviceIPhone::~AudioDeviceIPhone() { // API // ============================================================================ -void AudioDeviceIPhone::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) { +void AudioDeviceIOS::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -86,7 +88,7 @@ void AudioDeviceIPhone::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) { _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS); } -int32_t AudioDeviceIPhone::ActiveAudioLayer( +int32_t AudioDeviceIOS::ActiveAudioLayer( AudioDeviceModule::AudioLayer& audioLayer) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -94,7 +96,7 @@ int32_t AudioDeviceIPhone::ActiveAudioLayer( return 0; } -int32_t AudioDeviceIPhone::Init() { +int32_t AudioDeviceIOS::Init() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -137,7 +139,7 @@ int32_t AudioDeviceIPhone::Init() { return 0; } -int32_t AudioDeviceIPhone::Terminate() { +int32_t AudioDeviceIOS::Terminate() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -169,13 +171,13 @@ int32_t AudioDeviceIPhone::Terminate() { return 0; } -bool AudioDeviceIPhone::Initialized() const { +bool AudioDeviceIOS::Initialized() const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); return (_initialized); } -int32_t AudioDeviceIPhone::InitSpeaker() { +int32_t AudioDeviceIOS::InitSpeaker() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -205,7 +207,7 @@ int32_t AudioDeviceIPhone::InitSpeaker() { return 0; } -int32_t AudioDeviceIPhone::InitMicrophone() { +int32_t AudioDeviceIOS::InitMicrophone() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -236,19 +238,19 @@ int32_t AudioDeviceIPhone::InitMicrophone() { return 0; } -bool AudioDeviceIPhone::SpeakerIsInitialized() const { +bool AudioDeviceIOS::SpeakerIsInitialized() const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); return _speakerIsInitialized; } -bool AudioDeviceIPhone::MicrophoneIsInitialized() const { +bool AudioDeviceIOS::MicrophoneIsInitialized() const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); return _micIsInitialized; } -int32_t AudioDeviceIPhone::SpeakerVolumeIsAvailable(bool& available) { +int32_t AudioDeviceIOS::SpeakerVolumeIsAvailable(bool& available) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -257,16 +259,16 @@ int32_t AudioDeviceIPhone::SpeakerVolumeIsAvailable(bool& available) { return 0; } -int32_t AudioDeviceIPhone::SetSpeakerVolume(uint32_t volume) { +int32_t AudioDeviceIOS::SetSpeakerVolume(uint32_t volume) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, - "AudioDeviceIPhone::SetSpeakerVolume(volume=%u)", volume); + "AudioDeviceIOS::SetSpeakerVolume(volume=%u)", volume); WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " API call not supported on this platform"); return -1; } -int32_t AudioDeviceIPhone::SpeakerVolume(uint32_t& volume) const { +int32_t AudioDeviceIOS::SpeakerVolume(uint32_t& volume) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -276,13 +278,13 @@ int32_t AudioDeviceIPhone::SpeakerVolume(uint32_t& volume) const { } int32_t - AudioDeviceIPhone::SetWaveOutVolume(uint16_t volumeLeft, - uint16_t volumeRight) { + AudioDeviceIOS::SetWaveOutVolume(uint16_t volumeLeft, + uint16_t volumeRight) { WEBRTC_TRACE( kTraceModuleCall, kTraceAudioDevice, _id, - "AudioDeviceIPhone::SetWaveOutVolume(volumeLeft=%u, volumeRight=%u)", + "AudioDeviceIOS::SetWaveOutVolume(volumeLeft=%u, volumeRight=%u)", volumeLeft, volumeRight); WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, @@ -292,7 +294,7 @@ int32_t } int32_t -AudioDeviceIPhone::WaveOutVolume(uint16_t& /*volumeLeft*/, +AudioDeviceIOS::WaveOutVolume(uint16_t& /*volumeLeft*/, uint16_t& /*volumeRight*/) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -303,7 +305,7 @@ AudioDeviceIPhone::WaveOutVolume(uint16_t& /*volumeLeft*/, } int32_t - AudioDeviceIPhone::MaxSpeakerVolume(uint32_t& maxVolume) const { + AudioDeviceIOS::MaxSpeakerVolume(uint32_t& maxVolume) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -312,7 +314,7 @@ int32_t return -1; } -int32_t AudioDeviceIPhone::MinSpeakerVolume( +int32_t AudioDeviceIOS::MinSpeakerVolume( uint32_t& minVolume) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -323,7 +325,7 @@ int32_t AudioDeviceIPhone::MinSpeakerVolume( } int32_t - AudioDeviceIPhone::SpeakerVolumeStepSize(uint16_t& stepSize) const { + AudioDeviceIOS::SpeakerVolumeStepSize(uint16_t& stepSize) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -332,7 +334,7 @@ int32_t return -1; } -int32_t AudioDeviceIPhone::SpeakerMuteIsAvailable(bool& available) { +int32_t AudioDeviceIOS::SpeakerMuteIsAvailable(bool& available) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -341,7 +343,7 @@ int32_t AudioDeviceIPhone::SpeakerMuteIsAvailable(bool& available) { return 0; } -int32_t AudioDeviceIPhone::SetSpeakerMute(bool enable) { +int32_t AudioDeviceIOS::SetSpeakerMute(bool enable) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -350,7 +352,7 @@ int32_t AudioDeviceIPhone::SetSpeakerMute(bool enable) { return -1; } -int32_t AudioDeviceIPhone::SpeakerMute(bool& enabled) const { +int32_t AudioDeviceIOS::SpeakerMute(bool& enabled) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -359,7 +361,7 @@ int32_t AudioDeviceIPhone::SpeakerMute(bool& enabled) const { return -1; } -int32_t AudioDeviceIPhone::MicrophoneMuteIsAvailable(bool& available) { +int32_t AudioDeviceIOS::MicrophoneMuteIsAvailable(bool& available) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -368,7 +370,7 @@ int32_t AudioDeviceIPhone::MicrophoneMuteIsAvailable(bool& available) { return 0; } -int32_t AudioDeviceIPhone::SetMicrophoneMute(bool enable) { +int32_t AudioDeviceIOS::SetMicrophoneMute(bool enable) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -377,7 +379,7 @@ int32_t AudioDeviceIPhone::SetMicrophoneMute(bool enable) { return -1; } -int32_t AudioDeviceIPhone::MicrophoneMute(bool& enabled) const { +int32_t AudioDeviceIOS::MicrophoneMute(bool& enabled) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -386,7 +388,7 @@ int32_t AudioDeviceIPhone::MicrophoneMute(bool& enabled) const { return -1; } -int32_t AudioDeviceIPhone::MicrophoneBoostIsAvailable(bool& available) { +int32_t AudioDeviceIOS::MicrophoneBoostIsAvailable(bool& available) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -395,9 +397,9 @@ int32_t AudioDeviceIPhone::MicrophoneBoostIsAvailable(bool& available) { return 0; } -int32_t AudioDeviceIPhone::SetMicrophoneBoost(bool enable) { +int32_t AudioDeviceIOS::SetMicrophoneBoost(bool enable) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, - "AudioDeviceIPhone::SetMicrophoneBoost(enable=%u)", enable); + "AudioDeviceIOS::SetMicrophoneBoost(enable=%u)", enable); if (!_micIsInitialized) { WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, @@ -414,7 +416,7 @@ int32_t AudioDeviceIPhone::SetMicrophoneBoost(bool enable) { return 0; } -int32_t AudioDeviceIPhone::MicrophoneBoost(bool& enabled) const { +int32_t AudioDeviceIOS::MicrophoneBoost(bool& enabled) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); if (!_micIsInitialized) { @@ -428,7 +430,7 @@ int32_t AudioDeviceIPhone::MicrophoneBoost(bool& enabled) const { return 0; } -int32_t AudioDeviceIPhone::StereoRecordingIsAvailable(bool& available) { +int32_t AudioDeviceIOS::StereoRecordingIsAvailable(bool& available) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -437,9 +439,9 @@ int32_t AudioDeviceIPhone::StereoRecordingIsAvailable(bool& available) { return 0; } -int32_t AudioDeviceIPhone::SetStereoRecording(bool enable) { +int32_t AudioDeviceIOS::SetStereoRecording(bool enable) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, - "AudioDeviceIPhone::SetStereoRecording(enable=%u)", enable); + "AudioDeviceIOS::SetStereoRecording(enable=%u)", enable); if (enable) { WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, @@ -449,7 +451,7 @@ int32_t AudioDeviceIPhone::SetStereoRecording(bool enable) { return 0; } -int32_t AudioDeviceIPhone::StereoRecording(bool& enabled) const { +int32_t AudioDeviceIOS::StereoRecording(bool& enabled) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -457,7 +459,7 @@ int32_t AudioDeviceIPhone::StereoRecording(bool& enabled) const { return 0; } -int32_t AudioDeviceIPhone::StereoPlayoutIsAvailable(bool& available) { +int32_t AudioDeviceIOS::StereoPlayoutIsAvailable(bool& available) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -466,9 +468,9 @@ int32_t AudioDeviceIPhone::StereoPlayoutIsAvailable(bool& available) { return 0; } -int32_t AudioDeviceIPhone::SetStereoPlayout(bool enable) { +int32_t AudioDeviceIOS::SetStereoPlayout(bool enable) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, - "AudioDeviceIPhone::SetStereoPlayout(enable=%u)", enable); + "AudioDeviceIOS::SetStereoPlayout(enable=%u)", enable); if (enable) { WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, @@ -478,7 +480,7 @@ int32_t AudioDeviceIPhone::SetStereoPlayout(bool enable) { return 0; } -int32_t AudioDeviceIPhone::StereoPlayout(bool& enabled) const { +int32_t AudioDeviceIOS::StereoPlayout(bool& enabled) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -486,23 +488,23 @@ int32_t AudioDeviceIPhone::StereoPlayout(bool& enabled) const { return 0; } -int32_t AudioDeviceIPhone::SetAGC(bool enable) { +int32_t AudioDeviceIOS::SetAGC(bool enable) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, - "AudioDeviceIPhone::SetAGC(enable=%d)", enable); + "AudioDeviceIOS::SetAGC(enable=%d)", enable); _AGC = enable; return 0; } -bool AudioDeviceIPhone::AGC() const { +bool AudioDeviceIOS::AGC() const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); return _AGC; } -int32_t AudioDeviceIPhone::MicrophoneVolumeIsAvailable(bool& available) { +int32_t AudioDeviceIOS::MicrophoneVolumeIsAvailable(bool& available) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -511,9 +513,9 @@ int32_t AudioDeviceIPhone::MicrophoneVolumeIsAvailable(bool& available) { return 0; } -int32_t AudioDeviceIPhone::SetMicrophoneVolume(uint32_t volume) { +int32_t AudioDeviceIOS::SetMicrophoneVolume(uint32_t volume) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, - "AudioDeviceIPhone::SetMicrophoneVolume(volume=%u)", volume); + "AudioDeviceIOS::SetMicrophoneVolume(volume=%u)", volume); WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " API call not supported on this platform"); @@ -521,7 +523,7 @@ int32_t AudioDeviceIPhone::SetMicrophoneVolume(uint32_t volume) { } int32_t - AudioDeviceIPhone::MicrophoneVolume(uint32_t& volume) const { + AudioDeviceIOS::MicrophoneVolume(uint32_t& volume) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -531,7 +533,7 @@ int32_t } int32_t - AudioDeviceIPhone::MaxMicrophoneVolume(uint32_t& maxVolume) const { + AudioDeviceIOS::MaxMicrophoneVolume(uint32_t& maxVolume) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -541,7 +543,7 @@ int32_t } int32_t - AudioDeviceIPhone::MinMicrophoneVolume(uint32_t& minVolume) const { + AudioDeviceIOS::MinMicrophoneVolume(uint32_t& minVolume) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -551,7 +553,7 @@ int32_t } int32_t - AudioDeviceIPhone::MicrophoneVolumeStepSize( + AudioDeviceIOS::MicrophoneVolumeStepSize( uint16_t& stepSize) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -561,16 +563,16 @@ int32_t return -1; } -int16_t AudioDeviceIPhone::PlayoutDevices() { +int16_t AudioDeviceIOS::PlayoutDevices() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); return (int16_t)1; } -int32_t AudioDeviceIPhone::SetPlayoutDevice(uint16_t index) { +int32_t AudioDeviceIOS::SetPlayoutDevice(uint16_t index) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, - "AudioDeviceIPhone::SetPlayoutDevice(index=%u)", index); + "AudioDeviceIOS::SetPlayoutDevice(index=%u)", index); if (_playIsInitialized) { WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, @@ -589,18 +591,18 @@ int32_t AudioDeviceIPhone::SetPlayoutDevice(uint16_t index) { } int32_t - AudioDeviceIPhone::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType) { + AudioDeviceIOS::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType) { WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "WindowsDeviceType not supported"); return -1; } int32_t - AudioDeviceIPhone::PlayoutDeviceName(uint16_t index, + AudioDeviceIOS::PlayoutDeviceName(uint16_t index, char name[kAdmMaxDeviceNameSize], char guid[kAdmMaxGuidSize]) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, - "AudioDeviceIPhone::PlayoutDeviceName(index=%u)", index); + "AudioDeviceIOS::PlayoutDeviceName(index=%u)", index); if (index != 0) { return -1; @@ -615,11 +617,11 @@ int32_t } int32_t - AudioDeviceIPhone::RecordingDeviceName(uint16_t index, + AudioDeviceIOS::RecordingDeviceName(uint16_t index, char name[kAdmMaxDeviceNameSize], char guid[kAdmMaxGuidSize]) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, - "AudioDeviceIPhone::RecordingDeviceName(index=%u)", index); + "AudioDeviceIOS::RecordingDeviceName(index=%u)", index); if (index != 0) { return -1; @@ -633,15 +635,15 @@ int32_t return 0; } -int16_t AudioDeviceIPhone::RecordingDevices() { +int16_t AudioDeviceIOS::RecordingDevices() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); return (int16_t)1; } -int32_t AudioDeviceIPhone::SetRecordingDevice(uint16_t index) { +int32_t AudioDeviceIOS::SetRecordingDevice(uint16_t index) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, - "AudioDeviceIPhone::SetRecordingDevice(index=%u)", index); + "AudioDeviceIOS::SetRecordingDevice(index=%u)", index); if (_recIsInitialized) { WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, @@ -661,7 +663,7 @@ int32_t AudioDeviceIPhone::SetRecordingDevice(uint16_t index) { } int32_t - AudioDeviceIPhone::SetRecordingDevice( + AudioDeviceIOS::SetRecordingDevice( AudioDeviceModule::WindowsDeviceType) { WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WindowsDeviceType not supported"); @@ -671,52 +673,55 @@ int32_t // ---------------------------------------------------------------------------- // SetLoudspeakerStatus // -// Overrides the receiver playout route to speaker instead. See -// kAudioSessionProperty_OverrideCategoryDefaultToSpeaker in CoreAudio -// documentation. +// Change the default receiver playout route to speaker. +// // ---------------------------------------------------------------------------- -int32_t AudioDeviceIPhone::SetLoudspeakerStatus(bool enable) { +int32_t AudioDeviceIOS::SetLoudspeakerStatus(bool enable) { WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, - "AudioDeviceIPhone::SetLoudspeakerStatus(enable=%d)", enable); - - UInt32 doChangeDefaultRoute = enable ? 1 : 0; - OSStatus err = AudioSessionSetProperty( - kAudioSessionProperty_OverrideCategoryDefaultToSpeaker, - sizeof(doChangeDefaultRoute), &doChangeDefaultRoute); + "AudioDeviceIOS::SetLoudspeakerStatus(enable=%d)", enable); + + AVAudioSession* session = [AVAudioSession sharedInstance]; + NSString* category = session.category; + AVAudioSessionCategoryOptions options = session.categoryOptions; + // Respect old category options if category is + // AVAudioSessionCategoryPlayAndRecord. Otherwise reset it since old options + // might not be valid for this category. + if ([category isEqualToString:AVAudioSessionCategoryPlayAndRecord]) { + if (enable) { + options |= AVAudioSessionCategoryOptionDefaultToSpeaker; + } else { + options &= ~AVAudioSessionCategoryOptionDefaultToSpeaker; + } + } else { + options = AVAudioSessionCategoryOptionDefaultToSpeaker; + } - if (err != noErr) { - WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, - "Error changing default output route " \ - "(only available on iOS 3.1 or later)"); - return -1; + NSError* error = nil; + [session setCategory:AVAudioSessionCategoryPlayAndRecord + withOptions:options + error:&error]; + if (error != nil) { + WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, + "Error changing default output route "); + return -1; } return 0; } -int32_t AudioDeviceIPhone::GetLoudspeakerStatus(bool &enabled) const { +int32_t AudioDeviceIOS::GetLoudspeakerStatus(bool &enabled) const { WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, - "AudioDeviceIPhone::SetLoudspeakerStatus(enabled=?)"); - - UInt32 route(0); - UInt32 size = sizeof(route); - OSStatus err = AudioSessionGetProperty( - kAudioSessionProperty_OverrideCategoryDefaultToSpeaker, - &size, &route); - if (err != noErr) { - WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, - "Error changing default output route " \ - "(only available on iOS 3.1 or later)"); - return -1; - } + "AudioDeviceIOS::SetLoudspeakerStatus(enabled=?)"); - enabled = route == 1 ? true: false; + AVAudioSession* session = [AVAudioSession sharedInstance]; + AVAudioSessionCategoryOptions options = session.categoryOptions; + enabled = options & AVAudioSessionCategoryOptionDefaultToSpeaker; return 0; } -int32_t AudioDeviceIPhone::PlayoutIsAvailable(bool& available) { +int32_t AudioDeviceIOS::PlayoutIsAvailable(bool& available) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); available = false; @@ -734,7 +739,7 @@ int32_t AudioDeviceIPhone::PlayoutIsAvailable(bool& available) { return 0; } -int32_t AudioDeviceIPhone::RecordingIsAvailable(bool& available) { +int32_t AudioDeviceIOS::RecordingIsAvailable(bool& available) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); available = false; @@ -752,7 +757,7 @@ int32_t AudioDeviceIPhone::RecordingIsAvailable(bool& available) { return 0; } -int32_t AudioDeviceIPhone::InitPlayout() { +int32_t AudioDeviceIOS::InitPlayout() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); CriticalSectionScoped lock(&_critSect); @@ -803,12 +808,12 @@ int32_t AudioDeviceIPhone::InitPlayout() { return 0; } -bool AudioDeviceIPhone::PlayoutIsInitialized() const { +bool AudioDeviceIOS::PlayoutIsInitialized() const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); return (_playIsInitialized); } -int32_t AudioDeviceIPhone::InitRecording() { +int32_t AudioDeviceIOS::InitRecording() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); CriticalSectionScoped lock(&_critSect); @@ -861,12 +866,12 @@ int32_t AudioDeviceIPhone::InitRecording() { return 0; } -bool AudioDeviceIPhone::RecordingIsInitialized() const { +bool AudioDeviceIOS::RecordingIsInitialized() const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); return (_recIsInitialized); } -int32_t AudioDeviceIPhone::StartRecording() { +int32_t AudioDeviceIOS::StartRecording() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); CriticalSectionScoped lock(&_critSect); @@ -913,7 +918,7 @@ int32_t AudioDeviceIPhone::StartRecording() { return 0; } -int32_t AudioDeviceIPhone::StopRecording() { +int32_t AudioDeviceIOS::StopRecording() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); CriticalSectionScoped lock(&_critSect); @@ -937,12 +942,12 @@ int32_t AudioDeviceIPhone::StopRecording() { return 0; } -bool AudioDeviceIPhone::Recording() const { +bool AudioDeviceIOS::Recording() const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); return (_recording); } -int32_t AudioDeviceIPhone::StartPlayout() { +int32_t AudioDeviceIOS::StartPlayout() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); // This lock is (among other things) needed to avoid concurrency issues @@ -988,7 +993,7 @@ int32_t AudioDeviceIPhone::StartPlayout() { return 0; } -int32_t AudioDeviceIPhone::StopPlayout() { +int32_t AudioDeviceIOS::StopPlayout() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); CriticalSectionScoped lock(&_critSect); @@ -1012,7 +1017,7 @@ int32_t AudioDeviceIPhone::StopPlayout() { return 0; } -bool AudioDeviceIPhone::Playing() const { +bool AudioDeviceIOS::Playing() const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); return (_playing); @@ -1025,7 +1030,7 @@ bool AudioDeviceIPhone::Playing() const { // and set enable states after shutdown to same as current. // In capture thread audio device will be shutdown, then started again. // ---------------------------------------------------------------------------- -int32_t AudioDeviceIPhone::ResetAudioDevice() { +int32_t AudioDeviceIOS::ResetAudioDevice() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); CriticalSectionScoped lock(&_critSect); @@ -1067,22 +1072,21 @@ int32_t AudioDeviceIPhone::ResetAudioDevice() { return 0; } -int32_t AudioDeviceIPhone::PlayoutDelay(uint16_t& delayMS) const { +int32_t AudioDeviceIOS::PlayoutDelay(uint16_t& delayMS) const { delayMS = _playoutDelay; return 0; } -int32_t AudioDeviceIPhone::RecordingDelay(uint16_t& delayMS) const { +int32_t AudioDeviceIOS::RecordingDelay(uint16_t& delayMS) const { delayMS = _recordingDelay; return 0; } int32_t - AudioDeviceIPhone::SetPlayoutBuffer( - const AudioDeviceModule::BufferType type, - uint16_t sizeMS) { + AudioDeviceIOS::SetPlayoutBuffer(const AudioDeviceModule::BufferType type, + uint16_t sizeMS) { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, - "AudioDeviceIPhone::SetPlayoutBuffer(type=%u, sizeMS=%u)", + "AudioDeviceIOS::SetPlayoutBuffer(type=%u, sizeMS=%u)", type, sizeMS); WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, @@ -1091,7 +1095,7 @@ int32_t } int32_t - AudioDeviceIPhone::PlayoutBuffer(AudioDeviceModule::BufferType& type, + AudioDeviceIOS::PlayoutBuffer(AudioDeviceModule::BufferType& type, uint16_t& sizeMS) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -1102,7 +1106,7 @@ int32_t return 0; } -int32_t AudioDeviceIPhone::CPULoad(uint16_t& /*load*/) const { +int32_t AudioDeviceIOS::CPULoad(uint16_t& /*load*/) const { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, @@ -1110,35 +1114,35 @@ int32_t AudioDeviceIPhone::CPULoad(uint16_t& /*load*/) const { return -1; } -bool AudioDeviceIPhone::PlayoutWarning() const { +bool AudioDeviceIOS::PlayoutWarning() const { return (_playWarning > 0); } -bool AudioDeviceIPhone::PlayoutError() const { +bool AudioDeviceIOS::PlayoutError() const { return (_playError > 0); } -bool AudioDeviceIPhone::RecordingWarning() const { +bool AudioDeviceIOS::RecordingWarning() const { return (_recWarning > 0); } -bool AudioDeviceIPhone::RecordingError() const { +bool AudioDeviceIOS::RecordingError() const { return (_recError > 0); } -void AudioDeviceIPhone::ClearPlayoutWarning() { +void AudioDeviceIOS::ClearPlayoutWarning() { _playWarning = 0; } -void AudioDeviceIPhone::ClearPlayoutError() { +void AudioDeviceIOS::ClearPlayoutError() { _playError = 0; } -void AudioDeviceIPhone::ClearRecordingWarning() { +void AudioDeviceIOS::ClearRecordingWarning() { _recWarning = 0; } -void AudioDeviceIPhone::ClearRecordingError() { +void AudioDeviceIOS::ClearRecordingError() { _recError = 0; } @@ -1146,7 +1150,7 @@ void AudioDeviceIPhone::ClearRecordingError() { // Private Methods // ============================================================================ -int32_t AudioDeviceIPhone::InitPlayOrRecord() { +int32_t AudioDeviceIOS::InitPlayOrRecord() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); OSStatus result = -1; @@ -1187,18 +1191,32 @@ int32_t AudioDeviceIPhone::InitPlayOrRecord() { } // Set preferred hardware sample rate to 16 kHz - Float64 sampleRate(16000.0); - result = AudioSessionSetProperty( - kAudioSessionProperty_PreferredHardwareSampleRate, - sizeof(sampleRate), &sampleRate); - if (0 != result) { + NSError* error = nil; + AVAudioSession* session = [AVAudioSession sharedInstance]; + Float64 preferredSampleRate(16000.0); + [session setPreferredSampleRate:preferredSampleRate + error:&error]; + if (error != nil) { + const char* errorString = [[error localizedDescription] UTF8String]; WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, - "Could not set preferred sample rate (result=%d)", result); + "Could not set preferred sample rate: %s", errorString); + } + error = nil; + [session setMode:AVAudioSessionModeVoiceChat + error:&error]; + if (error != nil) { + const char* errorString = [[error localizedDescription] UTF8String]; + WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, + "Could not set mode: %s", errorString); + } + error = nil; + [session setCategory:AVAudioSessionCategoryPlayAndRecord + error:&error]; + if (error != nil) { + const char* errorString = [[error localizedDescription] UTF8String]; + WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, + "Could not set category: %s", errorString); } - - uint32_t voiceChat = kAudioSessionMode_VoiceChat; - AudioSessionSetProperty(kAudioSessionProperty_Mode, - sizeof(voiceChat), &voiceChat); ////////////////////// // Setup Voice Processing Audio Unit @@ -1293,7 +1311,7 @@ int32_t AudioDeviceIPhone::InitPlayOrRecord() { " Audio Unit playout opened in sampling rate %f", playoutDesc.mSampleRate); - playoutDesc.mSampleRate = sampleRate; + playoutDesc.mSampleRate = preferredSampleRate; // Store the sampling frequency to use towards the Audio Device Buffer // todo: Add 48 kHz (increase buffer sizes). Other fs? @@ -1362,7 +1380,7 @@ int32_t AudioDeviceIPhone::InitPlayOrRecord() { " Audio Unit recording opened in sampling rate %f", recordingDesc.mSampleRate); - recordingDesc.mSampleRate = sampleRate; + recordingDesc.mSampleRate = preferredSampleRate; // Set stream format for out/1 (use same sampling frequency as for in/1) recordingDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger @@ -1392,25 +1410,73 @@ int32_t AudioDeviceIPhone::InitPlayOrRecord() { } // Get hardware sample rate for logging (see if we get what we asked for) - Float64 hardwareSampleRate = 0.0; - size = sizeof(hardwareSampleRate); - result = AudioSessionGetProperty( - kAudioSessionProperty_CurrentHardwareSampleRate, &size, - &hardwareSampleRate); - if (0 != result) { - WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, - " Could not get current HW sample rate (result=%d)", result); - } + double sampleRate = session.sampleRate; WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " Current HW sample rate is %f, ADB sample rate is %d", - hardwareSampleRate, _adbSampFreq); + sampleRate, _adbSampFreq); + + // Listen to audio interruptions. + NSNotificationCenter* center = [NSNotificationCenter defaultCenter]; + id observer = + [center addObserverForName:AVAudioSessionInterruptionNotification + object:nil + queue:[NSOperationQueue mainQueue] + usingBlock:^(NSNotification* notification) { + NSNumber* typeNumber = + [notification userInfo][AVAudioSessionInterruptionTypeKey]; + AVAudioSessionInterruptionType type = + (AVAudioSessionInterruptionType)[typeNumber unsignedIntegerValue]; + switch (type) { + case AVAudioSessionInterruptionTypeBegan: + // At this point our audio session has been deactivated and the + // audio unit render callbacks no longer occur. Nothing to do. + break; + case AVAudioSessionInterruptionTypeEnded: { + NSError* error = nil; + AVAudioSession* session = [AVAudioSession sharedInstance]; + [session setActive:YES + error:&error]; + if (error != nil) { + WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, + "Error activating audio session"); + } + // Post interruption the audio unit render callbacks don't + // automatically continue, so we restart the unit manually here. + AudioOutputUnitStop(_auVoiceProcessing); + AudioOutputUnitStart(_auVoiceProcessing); + break; + } + } + }]; + // Increment refcount on observer using ARC bridge. Instance variable is a + // void* instead of an id because header is included in other pure C++ + // files. + _audioInterruptionObserver = (__bridge_retained void*)observer; + + // Activate audio session. + error = nil; + [session setActive:YES + error:&error]; + if (error != nil) { + WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, + "Error activating audio session"); + } return 0; } -int32_t AudioDeviceIPhone::ShutdownPlayOrRecord() { +int32_t AudioDeviceIOS::ShutdownPlayOrRecord() { WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__); + if (_audioInterruptionObserver != NULL) { + NSNotificationCenter* center = [NSNotificationCenter defaultCenter]; + // Transfer ownership of observer back to ARC, which will dealloc the + // observer once it exits this scope. + id observer = (__bridge_transfer id)_audioInterruptionObserver; + [center removeObserver:observer]; + _audioInterruptionObserver = NULL; + } + // Close and delete AU OSStatus result = -1; if (NULL != _auVoiceProcessing) { @@ -1435,13 +1501,13 @@ int32_t AudioDeviceIPhone::ShutdownPlayOrRecord() { // ============================================================================ OSStatus - AudioDeviceIPhone::RecordProcess(void *inRefCon, - AudioUnitRenderActionFlags *ioActionFlags, - const AudioTimeStamp *inTimeStamp, - UInt32 inBusNumber, - UInt32 inNumberFrames, - AudioBufferList *ioData) { - AudioDeviceIPhone* ptrThis = static_cast<AudioDeviceIPhone*>(inRefCon); + AudioDeviceIOS::RecordProcess(void *inRefCon, + AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *inTimeStamp, + UInt32 inBusNumber, + UInt32 inNumberFrames, + AudioBufferList *ioData) { + AudioDeviceIOS* ptrThis = static_cast<AudioDeviceIOS*>(inRefCon); return ptrThis->RecordProcessImpl(ioActionFlags, inTimeStamp, @@ -1451,11 +1517,10 @@ OSStatus OSStatus - AudioDeviceIPhone::RecordProcessImpl( - AudioUnitRenderActionFlags *ioActionFlags, - const AudioTimeStamp *inTimeStamp, - uint32_t inBusNumber, - uint32_t inNumberFrames) { + AudioDeviceIOS::RecordProcessImpl(AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *inTimeStamp, + uint32_t inBusNumber, + uint32_t inNumberFrames) { // Setup some basic stuff // Use temp buffer not to lock up recording buffer more than necessary // todo: Make dataTmp a member variable with static size that holds @@ -1561,20 +1626,20 @@ OSStatus } OSStatus - AudioDeviceIPhone::PlayoutProcess(void *inRefCon, - AudioUnitRenderActionFlags *ioActionFlags, - const AudioTimeStamp *inTimeStamp, - UInt32 inBusNumber, - UInt32 inNumberFrames, - AudioBufferList *ioData) { - AudioDeviceIPhone* ptrThis = static_cast<AudioDeviceIPhone*>(inRefCon); + AudioDeviceIOS::PlayoutProcess(void *inRefCon, + AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *inTimeStamp, + UInt32 inBusNumber, + UInt32 inNumberFrames, + AudioBufferList *ioData) { + AudioDeviceIOS* ptrThis = static_cast<AudioDeviceIOS*>(inRefCon); return ptrThis->PlayoutProcessImpl(inNumberFrames, ioData); } OSStatus - AudioDeviceIPhone::PlayoutProcessImpl(uint32_t inNumberFrames, - AudioBufferList *ioData) { + AudioDeviceIOS::PlayoutProcessImpl(uint32_t inNumberFrames, + AudioBufferList *ioData) { // Setup some basic stuff // assert(sizeof(short) == 2); // Assumption for implementation @@ -1677,7 +1742,7 @@ OSStatus return 0; } -void AudioDeviceIPhone::UpdatePlayoutDelay() { +void AudioDeviceIOS::UpdatePlayoutDelay() { ++_playoutDelayMeasurementCounter; if (_playoutDelayMeasurementCounter >= 100) { @@ -1686,36 +1751,25 @@ void AudioDeviceIPhone::UpdatePlayoutDelay() { // Since this is eventually rounded to integral ms, add 0.5ms // here to get round-to-nearest-int behavior instead of // truncation. - float totalDelaySeconds = 0.0005; + double totalDelaySeconds = 0.0005; // HW output latency - Float32 f32(0); - UInt32 size = sizeof(f32); - OSStatus result = AudioSessionGetProperty( - kAudioSessionProperty_CurrentHardwareOutputLatency, &size, &f32); - if (0 != result) { - WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, - "error HW latency (result=%d)", result); - } - assert(f32 >= 0); - totalDelaySeconds += f32; + AVAudioSession* session = [AVAudioSession sharedInstance]; + double latency = session.outputLatency; + assert(latency >= 0); + totalDelaySeconds += latency; // HW buffer duration - f32 = 0; - result = AudioSessionGetProperty( - kAudioSessionProperty_CurrentHardwareIOBufferDuration, &size, &f32); - if (0 != result) { - WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, - "error HW buffer duration (result=%d)", result); - } - assert(f32 >= 0); - totalDelaySeconds += f32; + double ioBufferDuration = session.IOBufferDuration; + assert(ioBufferDuration >= 0); + totalDelaySeconds += ioBufferDuration; // AU latency Float64 f64(0); - size = sizeof(f64); - result = AudioUnitGetProperty(_auVoiceProcessing, - kAudioUnitProperty_Latency, kAudioUnitScope_Global, 0, &f64, &size); + UInt32 size = sizeof(f64); + OSStatus result = AudioUnitGetProperty( + _auVoiceProcessing, kAudioUnitProperty_Latency, + kAudioUnitScope_Global, 0, &f64, &size); if (0 != result) { WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "error AU latency (result=%d)", result); @@ -1733,7 +1787,7 @@ void AudioDeviceIPhone::UpdatePlayoutDelay() { // todo: Add playout buffer? } -void AudioDeviceIPhone::UpdateRecordingDelay() { +void AudioDeviceIOS::UpdateRecordingDelay() { ++_recordingDelayMeasurementCounter; if (_recordingDelayMeasurementCounter >= 100) { @@ -1742,37 +1796,25 @@ void AudioDeviceIPhone::UpdateRecordingDelay() { // Since this is eventually rounded to integral ms, add 0.5ms // here to get round-to-nearest-int behavior instead of // truncation. - float totalDelaySeconds = 0.0005; + double totalDelaySeconds = 0.0005; // HW input latency - Float32 f32(0); - UInt32 size = sizeof(f32); - OSStatus result = AudioSessionGetProperty( - kAudioSessionProperty_CurrentHardwareInputLatency, &size, &f32); - if (0 != result) { - WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, - "error HW latency (result=%d)", result); - } - assert(f32 >= 0); - totalDelaySeconds += f32; + AVAudioSession* session = [AVAudioSession sharedInstance]; + double latency = session.inputLatency; + assert(latency >= 0); + totalDelaySeconds += latency; // HW buffer duration - f32 = 0; - result = AudioSessionGetProperty( - kAudioSessionProperty_CurrentHardwareIOBufferDuration, &size, &f32); - if (0 != result) { - WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, - "error HW buffer duration (result=%d)", result); - } - assert(f32 >= 0); - totalDelaySeconds += f32; + double ioBufferDuration = session.IOBufferDuration; + assert(ioBufferDuration >= 0); + totalDelaySeconds += ioBufferDuration; // AU latency Float64 f64(0); - size = sizeof(f64); - result = AudioUnitGetProperty(_auVoiceProcessing, - kAudioUnitProperty_Latency, - kAudioUnitScope_Global, 0, &f64, &size); + UInt32 size = sizeof(f64); + OSStatus result = AudioUnitGetProperty( + _auVoiceProcessing, kAudioUnitProperty_Latency, + kAudioUnitScope_Global, 0, &f64, &size); if (0 != result) { WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "error AU latency (result=%d)", result); @@ -1799,11 +1841,11 @@ void AudioDeviceIPhone::UpdateRecordingDelay() { } } -bool AudioDeviceIPhone::RunCapture(void* ptrThis) { - return static_cast<AudioDeviceIPhone*>(ptrThis)->CaptureWorkerThread(); +bool AudioDeviceIOS::RunCapture(void* ptrThis) { + return static_cast<AudioDeviceIOS*>(ptrThis)->CaptureWorkerThread(); } -bool AudioDeviceIPhone::CaptureWorkerThread() { +bool AudioDeviceIOS::CaptureWorkerThread() { if (_recording) { int bufPos = 0; unsigned int lowestSeq = 0; diff --git a/modules/audio_device/ios/audio_device_utility_ios.cc b/modules/audio_device/ios/audio_device_utility_ios.cc index 6bbceb92..33628175 100644 --- a/modules/audio_device/ios/audio_device_utility_ios.cc +++ b/modules/audio_device/ios/audio_device_utility_ios.cc @@ -15,7 +15,7 @@ #include "webrtc/system_wrappers/interface/trace.h" namespace webrtc { -AudioDeviceUtilityIPhone::AudioDeviceUtilityIPhone(const int32_t id) +AudioDeviceUtilityIOS::AudioDeviceUtilityIOS(const int32_t id) : _critSect(*CriticalSectionWrapper::CreateCriticalSection()), _id(id), @@ -24,15 +24,16 @@ AudioDeviceUtilityIPhone::AudioDeviceUtilityIPhone(const int32_t id) "%s created", __FUNCTION__); } -AudioDeviceUtilityIPhone::~AudioDeviceUtilityIPhone() { +AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS() { WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__); - CriticalSectionScoped lock(&_critSect); - + { + CriticalSectionScoped lock(&_critSect); + } delete &_critSect; } -int32_t AudioDeviceUtilityIPhone::Init() { +int32_t AudioDeviceUtilityIOS::Init() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); diff --git a/modules/audio_device/ios/audio_device_utility_ios.h b/modules/audio_device/ios/audio_device_utility_ios.h index 081ab82d..16948685 100644 --- a/modules/audio_device/ios/audio_device_utility_ios.h +++ b/modules/audio_device/ios/audio_device_utility_ios.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IPHONE_H -#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IPHONE_H +#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IOS_H +#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IOS_H #include "webrtc/modules/audio_device/audio_device_utility.h" #include "webrtc/modules/audio_device/include/audio_device.h" @@ -17,15 +17,15 @@ namespace webrtc { class CriticalSectionWrapper; -class AudioDeviceUtilityIPhone: public AudioDeviceUtility { -public: - AudioDeviceUtilityIPhone(const int32_t id); - AudioDeviceUtilityIPhone(); - virtual ~AudioDeviceUtilityIPhone(); +class AudioDeviceUtilityIOS: public AudioDeviceUtility { + public: + AudioDeviceUtilityIOS(const int32_t id); + AudioDeviceUtilityIOS(); + virtual ~AudioDeviceUtilityIOS(); virtual int32_t Init(); -private: + private: CriticalSectionWrapper& _critSect; int32_t _id; AudioDeviceModule::ErrorCode _lastError; @@ -33,4 +33,4 @@ private: } // namespace webrtc -#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IPHONE_H +#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IOS_H diff --git a/modules/audio_processing/aecm/aecm_core_mips.c b/modules/audio_processing/aecm/aecm_core_mips.c index 31f232b8..4c925ca2 100644 --- a/modules/audio_processing/aecm/aecm_core_mips.c +++ b/modules/audio_processing/aecm/aecm_core_mips.c @@ -1485,7 +1485,7 @@ static void ComfortNoise(AecmCore_t* aecm, "mul %[tmp321], %[tmp321], %[tmp161] \n\t" "sra %[nrsh1], %[tmp32], 14 \n\t" "sra %[nrsh2], %[tmp321], 14 \n\t" - : [nrsh1] "=r" (nrsh1), [nrsh2] "=r" (nrsh2) + : [nrsh1] "=&r" (nrsh1), [nrsh2] "=r" (nrsh2) : [tmp16] "r" (tmp16), [tmp161] "r" (tmp161), [tmp32] "r" (tmp32), [tmp321] "r" (tmp321) : "memory", "hi", "lo" diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc index 35e1eb7c..fb2c200e 100644 --- a/modules/audio_processing/audio_buffer.cc +++ b/modules/audio_processing/audio_buffer.cc @@ -51,7 +51,6 @@ int KeyboardChannelIndex(AudioProcessing::ChannelLayout layout) { return -1; } - void StereoToMono(const float* left, const float* right, float* out, int samples_per_channel) { for (int i = 0; i < samples_per_channel; ++i) { @@ -71,8 +70,9 @@ void StereoToMono(const int16_t* left, const int16_t* right, int16_t* out, // One int16_t and one float ChannelBuffer that are kept in sync. The sync is // broken when someone requests write access to either ChannelBuffer, and // reestablished when someone requests the outdated ChannelBuffer. It is -// therefore safe to use the return value of ibuf() and fbuf() until the next -// call to the other method. +// therefore safe to use the return value of ibuf_const() and fbuf_const() +// until the next call to ibuf() or fbuf(), and the return value of ibuf() and +// fbuf() until the next call to any of the other functions. class IFChannelBuffer { public: IFChannelBuffer(int samples_per_channel, int num_channels) @@ -81,19 +81,24 @@ class IFChannelBuffer { fvalid_(true), fbuf_(samples_per_channel, num_channels) {} - ChannelBuffer<int16_t>* ibuf() { + ChannelBuffer<int16_t>* ibuf() { return ibuf(false); } + ChannelBuffer<float>* fbuf() { return fbuf(false); } + const ChannelBuffer<int16_t>* ibuf_const() { return ibuf(true); } + const ChannelBuffer<float>* fbuf_const() { return fbuf(true); } + + private: + ChannelBuffer<int16_t>* ibuf(bool readonly) { RefreshI(); - fvalid_ = false; + fvalid_ = readonly; return &ibuf_; } - ChannelBuffer<float>* fbuf() { + ChannelBuffer<float>* fbuf(bool readonly) { RefreshF(); - ivalid_ = false; + ivalid_ = readonly; return &fbuf_; } - private: void RefreshF() { if (!fvalid_) { assert(ivalid_); @@ -126,24 +131,6 @@ class IFChannelBuffer { ChannelBuffer<float> fbuf_; }; -class SplitChannelBuffer { - public: - SplitChannelBuffer(int samples_per_split_channel, int num_channels) - : low_(samples_per_split_channel, num_channels), - high_(samples_per_split_channel, num_channels) { - } - ~SplitChannelBuffer() {} - - int16_t* low_channel(int i) { return low_.ibuf()->channel(i); } - int16_t* high_channel(int i) { return high_.ibuf()->channel(i); } - float* low_channel_f(int i) { return low_.fbuf()->channel(i); } - float* high_channel_f(int i) { return high_.fbuf()->channel(i); } - - private: - IFChannelBuffer low_; - IFChannelBuffer high_; -}; - AudioBuffer::AudioBuffer(int input_samples_per_channel, int num_input_channels, int process_samples_per_channel, @@ -155,8 +142,7 @@ AudioBuffer::AudioBuffer(int input_samples_per_channel, num_proc_channels_(num_process_channels), output_samples_per_channel_(output_samples_per_channel), samples_per_split_channel_(proc_samples_per_channel_), - num_mixed_channels_(0), - num_mixed_low_pass_channels_(0), + mixed_low_pass_valid_(false), reference_copied_(false), activity_(AudioFrame::kVadUnknown), keyboard_data_(NULL), @@ -200,8 +186,10 @@ AudioBuffer::AudioBuffer(int input_samples_per_channel, if (proc_samples_per_channel_ == kSamplesPer32kHzChannel) { samples_per_split_channel_ = kSamplesPer16kHzChannel; - split_channels_.reset(new SplitChannelBuffer(samples_per_split_channel_, - num_proc_channels_)); + split_channels_low_.reset(new IFChannelBuffer(samples_per_split_channel_, + num_proc_channels_)); + split_channels_high_.reset(new IFChannelBuffer(samples_per_split_channel_, + num_proc_channels_)); filter_states_.reset(new SplitFilterStates[num_proc_channels_]); } } @@ -278,89 +266,102 @@ void AudioBuffer::CopyTo(int samples_per_channel, void AudioBuffer::InitForNewData() { keyboard_data_ = NULL; - num_mixed_channels_ = 0; - num_mixed_low_pass_channels_ = 0; + mixed_low_pass_valid_ = false; reference_copied_ = false; activity_ = AudioFrame::kVadUnknown; } const int16_t* AudioBuffer::data(int channel) const { - assert(channel >= 0 && channel < num_proc_channels_); - return channels_->ibuf()->channel(channel); + return channels_->ibuf_const()->channel(channel); } int16_t* AudioBuffer::data(int channel) { - const AudioBuffer* t = this; - return const_cast<int16_t*>(t->data(channel)); + mixed_low_pass_valid_ = false; + return channels_->ibuf()->channel(channel); } const float* AudioBuffer::data_f(int channel) const { - assert(channel >= 0 && channel < num_proc_channels_); - return channels_->fbuf()->channel(channel); + return channels_->fbuf_const()->channel(channel); } float* AudioBuffer::data_f(int channel) { - const AudioBuffer* t = this; - return const_cast<float*>(t->data_f(channel)); + mixed_low_pass_valid_ = false; + return channels_->fbuf()->channel(channel); } const int16_t* AudioBuffer::low_pass_split_data(int channel) const { - assert(channel >= 0 && channel < num_proc_channels_); - return split_channels_.get() ? split_channels_->low_channel(channel) - : data(channel); + return split_channels_low_.get() + ? split_channels_low_->ibuf_const()->channel(channel) + : data(channel); } int16_t* AudioBuffer::low_pass_split_data(int channel) { - const AudioBuffer* t = this; - return const_cast<int16_t*>(t->low_pass_split_data(channel)); + mixed_low_pass_valid_ = false; + return split_channels_low_.get() + ? split_channels_low_->ibuf()->channel(channel) + : data(channel); } const float* AudioBuffer::low_pass_split_data_f(int channel) const { - assert(channel >= 0 && channel < num_proc_channels_); - return split_channels_.get() ? split_channels_->low_channel_f(channel) - : data_f(channel); + return split_channels_low_.get() + ? split_channels_low_->fbuf_const()->channel(channel) + : data_f(channel); } float* AudioBuffer::low_pass_split_data_f(int channel) { - const AudioBuffer* t = this; - return const_cast<float*>(t->low_pass_split_data_f(channel)); + mixed_low_pass_valid_ = false; + return split_channels_low_.get() + ? split_channels_low_->fbuf()->channel(channel) + : data_f(channel); } const int16_t* AudioBuffer::high_pass_split_data(int channel) const { - assert(channel >= 0 && channel < num_proc_channels_); - return split_channels_.get() ? split_channels_->high_channel(channel) : NULL; + return split_channels_high_.get() + ? split_channels_high_->ibuf_const()->channel(channel) + : NULL; } int16_t* AudioBuffer::high_pass_split_data(int channel) { - const AudioBuffer* t = this; - return const_cast<int16_t*>(t->high_pass_split_data(channel)); + return split_channels_high_.get() + ? split_channels_high_->ibuf()->channel(channel) + : NULL; } const float* AudioBuffer::high_pass_split_data_f(int channel) const { - assert(channel >= 0 && channel < num_proc_channels_); - return split_channels_.get() ? split_channels_->high_channel_f(channel) - : NULL; + return split_channels_high_.get() + ? split_channels_high_->fbuf_const()->channel(channel) + : NULL; } float* AudioBuffer::high_pass_split_data_f(int channel) { - const AudioBuffer* t = this; - return const_cast<float*>(t->high_pass_split_data_f(channel)); + return split_channels_high_.get() + ? split_channels_high_->fbuf()->channel(channel) + : NULL; } -const int16_t* AudioBuffer::mixed_data(int channel) const { - assert(channel >= 0 && channel < num_mixed_channels_); - - return mixed_channels_->channel(channel); -} +const int16_t* AudioBuffer::mixed_low_pass_data() { + // Currently only mixing stereo to mono is supported. + assert(num_proc_channels_ == 1 || num_proc_channels_ == 2); -const int16_t* AudioBuffer::mixed_low_pass_data(int channel) const { - assert(channel >= 0 && channel < num_mixed_low_pass_channels_); + if (num_proc_channels_ == 1) { + return low_pass_split_data(0); + } - return mixed_low_pass_channels_->channel(channel); + if (!mixed_low_pass_valid_) { + if (!mixed_low_pass_channels_.get()) { + mixed_low_pass_channels_.reset( + new ChannelBuffer<int16_t>(samples_per_split_channel_, 1)); + } + StereoToMono(low_pass_split_data(0), + low_pass_split_data(1), + mixed_low_pass_channels_->data(), + samples_per_split_channel_); + mixed_low_pass_valid_ = true; + } + return mixed_low_pass_channels_->data(); } const int16_t* AudioBuffer::low_pass_reference(int channel) const { - assert(channel >= 0 && channel < num_proc_channels_); if (!reference_copied_) { return NULL; } @@ -444,42 +445,6 @@ void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const { } } -void AudioBuffer::CopyAndMix(int num_mixed_channels) { - // We currently only support the stereo to mono case. - assert(num_proc_channels_ == 2); - assert(num_mixed_channels == 1); - if (!mixed_channels_.get()) { - mixed_channels_.reset( - new ChannelBuffer<int16_t>(proc_samples_per_channel_, - num_mixed_channels)); - } - - StereoToMono(channels_->ibuf()->channel(0), - channels_->ibuf()->channel(1), - mixed_channels_->channel(0), - proc_samples_per_channel_); - - num_mixed_channels_ = num_mixed_channels; -} - -void AudioBuffer::CopyAndMixLowPass(int num_mixed_channels) { - // We currently only support the stereo to mono case. - assert(num_proc_channels_ == 2); - assert(num_mixed_channels == 1); - if (!mixed_low_pass_channels_.get()) { - mixed_low_pass_channels_.reset( - new ChannelBuffer<int16_t>(samples_per_split_channel_, - num_mixed_channels)); - } - - StereoToMono(low_pass_split_data(0), - low_pass_split_data(1), - mixed_low_pass_channels_->channel(0), - samples_per_split_channel_); - - num_mixed_low_pass_channels_ = num_mixed_channels; -} - void AudioBuffer::CopyLowPassToReference() { reference_copied_ = true; if (!low_pass_reference_channels_.get()) { diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h index db24e959..acf5753c 100644 --- a/modules/audio_processing/audio_buffer.h +++ b/modules/audio_processing/audio_buffer.h @@ -23,7 +23,6 @@ namespace webrtc { class PushSincResampler; -class SplitChannelBuffer; class IFChannelBuffer; struct SplitFilterStates { @@ -56,15 +55,18 @@ class AudioBuffer { int samples_per_split_channel() const; int samples_per_keyboard_channel() const; - // It can be assumed that channels are stored contiguously. + // Sample array accessors. Channels are guaranteed to be stored contiguously + // in memory. Prefer to use the const variants of each accessor when + // possible, since they incur less float<->int16 conversion overhead. int16_t* data(int channel); const int16_t* data(int channel) const; int16_t* low_pass_split_data(int channel); const int16_t* low_pass_split_data(int channel) const; int16_t* high_pass_split_data(int channel); const int16_t* high_pass_split_data(int channel) const; - const int16_t* mixed_data(int channel) const; - const int16_t* mixed_low_pass_data(int channel) const; + // Returns a pointer to the low-pass data downmixed to mono. If this data + // isn't already available it re-calculates it. + const int16_t* mixed_low_pass_data(); const int16_t* low_pass_reference(int channel) const; // Float versions of the accessors, with automatic conversion back and forth @@ -85,7 +87,6 @@ class AudioBuffer { // Use for int16 interleaved data. void DeinterleaveFrom(AudioFrame* audioFrame); - void InterleaveTo(AudioFrame* audioFrame) const; // If |data_changed| is false, only the non-audio data members will be copied // to |frame|. void InterleaveTo(AudioFrame* frame, bool data_changed) const; @@ -97,9 +98,6 @@ class AudioBuffer { void CopyTo(int samples_per_channel, AudioProcessing::ChannelLayout layout, float* const* data); - - void CopyAndMix(int num_mixed_channels); - void CopyAndMixLowPass(int num_mixed_channels); void CopyLowPassToReference(); private: @@ -112,16 +110,15 @@ class AudioBuffer { const int num_proc_channels_; const int output_samples_per_channel_; int samples_per_split_channel_; - int num_mixed_channels_; - int num_mixed_low_pass_channels_; + bool mixed_low_pass_valid_; bool reference_copied_; AudioFrame::VADActivity activity_; const float* keyboard_data_; scoped_ptr<IFChannelBuffer> channels_; - scoped_ptr<SplitChannelBuffer> split_channels_; + scoped_ptr<IFChannelBuffer> split_channels_low_; + scoped_ptr<IFChannelBuffer> split_channels_high_; scoped_ptr<SplitFilterStates[]> filter_states_; - scoped_ptr<ChannelBuffer<int16_t> > mixed_channels_; scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_; scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_; scoped_ptr<ChannelBuffer<float> > input_buffer_; diff --git a/modules/audio_processing/common.h b/modules/audio_processing/common.h index 42454df2..98e36cb0 100644 --- a/modules/audio_processing/common.h +++ b/modules/audio_processing/common.h @@ -54,10 +54,14 @@ class ChannelBuffer { } T* data() { return data_.get(); } - T* channel(int i) { - assert(i < num_channels_); + const T* channel(int i) const { + assert(i >= 0 && i < num_channels_); return channels_[i]; } + T* channel(int i) { + const ChannelBuffer<T>* t = this; + return const_cast<T*>(t->channel(i)); + } T** channels() { return channels_.get(); } int samples_per_channel() { return samples_per_channel_; } diff --git a/modules/audio_processing/gain_control_impl.cc b/modules/audio_processing/gain_control_impl.cc index a67b67ec..cf7df169 100644 --- a/modules/audio_processing/gain_control_impl.cc +++ b/modules/audio_processing/gain_control_impl.cc @@ -59,17 +59,11 @@ int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) { assert(audio->samples_per_split_channel() <= 160); - const int16_t* mixed_data = audio->low_pass_split_data(0); - if (audio->num_channels() > 1) { - audio->CopyAndMixLowPass(1); - mixed_data = audio->mixed_low_pass_data(0); - } - for (int i = 0; i < num_handles(); i++) { Handle* my_handle = static_cast<Handle*>(handle(i)); int err = WebRtcAgc_AddFarend( my_handle, - mixed_data, + audio->mixed_low_pass_data(), static_cast<int16_t>(audio->samples_per_split_channel())); if (err != apm_->kNoError) { diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h index 6f4cc9e7..30f0d9c5 100644 --- a/modules/audio_processing/include/audio_processing.h +++ b/modules/audio_processing/include/audio_processing.h @@ -209,9 +209,6 @@ class AudioProcessing { // ensures the options are applied immediately. virtual void SetExtraOptions(const Config& config) = 0; - virtual int EnableExperimentalNs(bool enable) { return kNoError; } - virtual bool experimental_ns_enabled() const { return false; } - // DEPRECATED. // TODO(ajm): Remove after Chromium has upgraded to using Initialize(). virtual int set_sample_rate_hz(int rate) = 0; diff --git a/modules/audio_processing/utility/ring_buffer_unittest.cc b/modules/audio_processing/utility/ring_buffer_unittest.cc index 5dacf0b8..f5c36c2a 100644 --- a/modules/audio_processing/utility/ring_buffer_unittest.cc +++ b/modules/audio_processing/utility/ring_buffer_unittest.cc @@ -52,8 +52,8 @@ static int CheckIncrementingData(int* data, int num_elements, // We use ASSERTs in this test to avoid obscuring the seed in the case of a // failure. static void RandomStressTest(int** data_ptr) { - const int kNumTests = 100; - const int kNumOps = 10000; + const int kNumTests = 10; + const int kNumOps = 1000; const int kMaxBufferSize = 1000; unsigned int seed = time(NULL); diff --git a/modules/audio_processing/voice_detection_impl.cc b/modules/audio_processing/voice_detection_impl.cc index c6e497ff..31336b41 100644 --- a/modules/audio_processing/voice_detection_impl.cc +++ b/modules/audio_processing/voice_detection_impl.cc @@ -61,17 +61,11 @@ int VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) { } assert(audio->samples_per_split_channel() <= 160); - const int16_t* mixed_data = audio->low_pass_split_data(0); - if (audio->num_channels() > 1) { - audio->CopyAndMixLowPass(1); - mixed_data = audio->mixed_low_pass_data(0); - } - // TODO(ajm): concatenate data in frame buffer here. int vad_ret = WebRtcVad_Process(static_cast<Handle*>(handle(0)), apm_->proc_split_sample_rate_hz(), - mixed_data, + audio->mixed_low_pass_data(), frame_size_samples_); if (vad_ret == 0) { stream_has_voice_ = false; diff --git a/modules/desktop_capture/win/cursor.cc b/modules/desktop_capture/win/cursor.cc index 00055c44..e3c272ce 100644 --- a/modules/desktop_capture/win/cursor.cc +++ b/modules/desktop_capture/win/cursor.cc @@ -197,7 +197,7 @@ MouseCursor* CreateMouseCursorFromHCursor(HDC dc, HCURSOR cursor) { // The XOR mask becomes the color bitmap. memcpy( - image->data(), mask_plane + (width * height), image->stride() * width); + image->data(), mask_plane + (width * height), image->stride() * height); } // Reconstruct transparency from the mask if the color image does not has diff --git a/modules/media_file/source/media_file_unittest.cc b/modules/media_file/source/media_file_unittest.cc index d658dc2c..56d3544c 100644 --- a/modules/media_file/source/media_file_unittest.cc +++ b/modules/media_file/source/media_file_unittest.cc @@ -10,6 +10,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/modules/media_file/interface/media_file.h" +#include "webrtc/system_wrappers/interface/compile_assert.h" #include "webrtc/system_wrappers/interface/sleep.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/gtest_disable.h" @@ -45,3 +46,50 @@ TEST_F(MediaFileTest, DISABLED_ON_ANDROID(StartPlayingAudioFileWithoutError)) { ASSERT_EQ(0, media_file_->StopPlaying()); } + +TEST_F(MediaFileTest, WriteWavFile) { + // Write file. + static const int kHeaderSize = 44; + static const int kPayloadSize = 320; + webrtc::CodecInst codec = {0, "L16", 16000, kPayloadSize, 1}; + std::string outfile = webrtc::test::OutputPath() + "wavtest.wav"; + ASSERT_EQ(0, + media_file_->StartRecordingAudioFile( + outfile.c_str(), webrtc::kFileFormatWavFile, codec)); + static const int8_t kFakeData[kPayloadSize] = {0}; + ASSERT_EQ(0, media_file_->IncomingAudioData(kFakeData, kPayloadSize)); + ASSERT_EQ(0, media_file_->StopRecording()); + + // Check the file we just wrote. + static const uint8_t kExpectedHeader[] = { + 'R', 'I', 'F', 'F', + 0x64, 0x1, 0, 0, // size of whole file - 8: 320 + 44 - 8 + 'W', 'A', 'V', 'E', + 'f', 'm', 't', ' ', + 0x10, 0, 0, 0, // size of fmt block - 8: 24 - 8 + 0x1, 0, // format: PCM (1) + 0x1, 0, // channels: 1 + 0x80, 0x3e, 0, 0, // sample rate: 16000 + 0, 0x7d, 0, 0, // byte rate: 2 * 16000 + 0x2, 0, // block align: NumChannels * BytesPerSample + 0x10, 0, // bits per sample: 2 * 8 + 'd', 'a', 't', 'a', + 0x40, 0x1, 0, 0, // size of payload: 320 + }; + COMPILE_ASSERT(sizeof(kExpectedHeader) == kHeaderSize, header_size); + + EXPECT_EQ(size_t(kHeaderSize + kPayloadSize), + webrtc::test::GetFileSize(outfile)); + FILE* f = fopen(outfile.c_str(), "rb"); + ASSERT_TRUE(f); + + uint8_t header[kHeaderSize]; + ASSERT_EQ(1u, fread(header, kHeaderSize, 1, f)); + EXPECT_EQ(0, memcmp(kExpectedHeader, header, kHeaderSize)); + + uint8_t payload[kPayloadSize]; + ASSERT_EQ(1u, fread(payload, kPayloadSize, 1, f)); + EXPECT_EQ(0, memcmp(kFakeData, payload, kPayloadSize)); + + EXPECT_EQ(0, fclose(f)); +} diff --git a/modules/modules.gyp b/modules/modules.gyp index d054fe9e..2a3ba743 100644 --- a/modules/modules.gyp +++ b/modules/modules.gyp @@ -249,7 +249,6 @@ 'video_processing/main/test/unit_test/color_enhancement_test.cc', 'video_processing/main/test/unit_test/content_metrics_test.cc', 'video_processing/main/test/unit_test/deflickering_test.cc', - 'video_processing/main/test/unit_test/denoising_test.cc', 'video_processing/main/test/unit_test/video_processing_unittest.cc', 'video_processing/main/test/unit_test/video_processing_unittest.h', ], diff --git a/modules/pacing/include/paced_sender.h b/modules/pacing/include/paced_sender.h index b9151a5f..ddd8e53b 100644 --- a/modules/pacing/include/paced_sender.h +++ b/modules/pacing/include/paced_sender.h @@ -146,8 +146,8 @@ class PacedSender : public Module { scoped_ptr<paced_sender::IntervalBudget> padding_budget_ GUARDED_BY(critsect_); - int64_t time_last_update_ GUARDED_BY(critsect_); - int64_t time_last_send_ GUARDED_BY(critsect_); + int64_t time_last_update_us_ GUARDED_BY(critsect_); + int64_t time_last_send_us_ GUARDED_BY(critsect_); int64_t capture_time_ms_last_queued_ GUARDED_BY(critsect_); int64_t capture_time_ms_last_sent_ GUARDED_BY(critsect_); diff --git a/modules/pacing/paced_sender.cc b/modules/pacing/paced_sender.cc index 52e9cfb4..6204a9a0 100644 --- a/modules/pacing/paced_sender.cc +++ b/modules/pacing/paced_sender.cc @@ -31,12 +31,11 @@ const int kMaxIntervalTimeMs = 30; // Max time that the first packet in the queue can sit in the queue if no // packets are sent, regardless of buffer state. In practice only in effect at // low bitrates (less than 320 kbits/s). -const int kMaxQueueTimeWithoutSendingMs = 30; +const int kMaxQueueTimeWithoutSendingUs = 30000; } // namespace namespace webrtc { - namespace paced_sender { struct Packet { Packet(uint32_t ssrc, @@ -142,7 +141,7 @@ PacedSender::PacedSender(Clock* clock, max_queue_length_ms_(kDefaultMaxQueueLengthMs), media_budget_(new paced_sender::IntervalBudget(max_bitrate_kbps)), padding_budget_(new paced_sender::IntervalBudget(min_bitrate_kbps)), - time_last_update_(clock->TimeInMilliseconds()), + time_last_update_us_(clock->TimeInMicroseconds()), capture_time_ms_last_queued_(0), capture_time_ms_last_sent_(0), high_priority_packets_(new paced_sender::PacketList), @@ -151,8 +150,7 @@ PacedSender::PacedSender(Clock* clock, UpdateBytesPerInterval(kMinPacketLimitMs); } -PacedSender::~PacedSender() { -} +PacedSender::~PacedSender() {} void PacedSender::Pause() { CriticalSectionScoped cs(critsect_.get()); @@ -248,7 +246,8 @@ int PacedSender::QueueInMs() const { int32_t PacedSender::TimeUntilNextProcess() { CriticalSectionScoped cs(critsect_.get()); - int64_t elapsed_time_ms = clock_->TimeInMilliseconds() - time_last_update_; + int64_t elapsed_time_ms = (clock_->TimeInMicroseconds() - + time_last_update_us_ + 500) / 1000; if (elapsed_time_ms <= 0) { return kMinPacketLimitMs; } @@ -259,10 +258,10 @@ int32_t PacedSender::TimeUntilNextProcess() { } int32_t PacedSender::Process() { - int64_t now = clock_->TimeInMilliseconds(); + int64_t now_us = clock_->TimeInMicroseconds(); CriticalSectionScoped cs(critsect_.get()); - int elapsed_time_ms = now - time_last_update_; - time_last_update_ = now; + int elapsed_time_ms = (now_us - time_last_update_us_ + 500) / 1000; + time_last_update_us_ = now_us; if (!enabled_) { return 0; } @@ -291,7 +290,6 @@ int32_t PacedSender::Process() { return 0; } -// MUST have critsect_ when calling. bool PacedSender::SendPacketFromList(paced_sender::PacketList* packet_list) EXCLUSIVE_LOCKS_REQUIRED(critsect_.get()) { paced_sender::Packet packet = GetNextPacketFromList(packet_list); @@ -322,20 +320,18 @@ bool PacedSender::SendPacketFromList(paced_sender::PacketList* packet_list) return true; } -// MUST have critsect_ when calling. void PacedSender::UpdateBytesPerInterval(uint32_t delta_time_ms) { media_budget_->IncreaseBudget(delta_time_ms); padding_budget_->IncreaseBudget(delta_time_ms); } -// MUST have critsect_ when calling. bool PacedSender::ShouldSendNextPacket(paced_sender::PacketList** packet_list) { *packet_list = NULL; if (media_budget_->bytes_remaining() <= 0) { // All bytes consumed for this interval. // Check if we have not sent in a too long time. - if (clock_->TimeInMilliseconds() - time_last_send_ > - kMaxQueueTimeWithoutSendingMs) { + if (clock_->TimeInMicroseconds() - time_last_send_us_ > + kMaxQueueTimeWithoutSendingUs) { if (!high_priority_packets_->empty()) { *packet_list = high_priority_packets_.get(); return true; @@ -386,9 +382,8 @@ paced_sender::Packet PacedSender::GetNextPacketFromList( return packet; } -// MUST have critsect_ when calling. void PacedSender::UpdateMediaBytesSent(int num_bytes) { - time_last_send_ = clock_->TimeInMilliseconds(); + time_last_send_us_ = clock_->TimeInMicroseconds(); media_budget_->UseBudget(num_bytes); padding_budget_->UseBudget(num_bytes); } diff --git a/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/modules/rtp_rtcp/interface/rtp_rtcp_defines.h index e1bec5fc..99808439 100644 --- a/modules/rtp_rtcp/interface/rtp_rtcp_defines.h +++ b/modules/rtp_rtcp/interface/rtp_rtcp_defines.h @@ -198,12 +198,14 @@ struct RtpState { start_timestamp(0), timestamp(0), capture_time_ms(-1), - last_timestamp_time_ms(-1) {} + last_timestamp_time_ms(-1), + media_has_been_sent(false) {} uint16_t sequence_number; uint32_t start_timestamp; uint32_t timestamp; int64_t capture_time_ms; int64_t last_timestamp_time_ms; + bool media_has_been_sent; }; class RtpData diff --git a/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc index 0514277f..5e580a34 100644 --- a/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc +++ b/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc @@ -121,7 +121,8 @@ TEST_F(RtcpFormatRembTest, TestNonCompund) { uint32_t SSRC = 456789; EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpNonCompound)); EXPECT_EQ(0, rtcp_sender_->SetREMBData(1234, 1, &SSRC)); - RTCPSender::FeedbackState feedback_state(dummy_rtp_rtcp_impl_); + RTCPSender::FeedbackState feedback_state = + dummy_rtp_rtcp_impl_->GetFeedbackState(); EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpRemb)); } @@ -129,7 +130,8 @@ TEST_F(RtcpFormatRembTest, TestCompund) { uint32_t SSRCs[2] = {456789, 98765}; EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpCompound)); EXPECT_EQ(0, rtcp_sender_->SetREMBData(1234, 2, SSRCs)); - RTCPSender::FeedbackState feedback_state(dummy_rtp_rtcp_impl_); + RTCPSender::FeedbackState feedback_state = + dummy_rtp_rtcp_impl_->GetFeedbackState(); EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpRemb)); } } // namespace diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc index b38ae1f0..54b991bf 100644 --- a/modules/rtp_rtcp/source/rtcp_receiver.cc +++ b/modules/rtp_rtcp/source/rtcp_receiver.cc @@ -224,12 +224,12 @@ bool RTCPReceiver::GetAndResetXrRrRtt(uint16_t* rtt_ms) { return true; } -int32_t -RTCPReceiver::NTP(uint32_t *ReceivedNTPsecs, - uint32_t *ReceivedNTPfrac, - uint32_t *RTCPArrivalTimeSecs, - uint32_t *RTCPArrivalTimeFrac, - uint32_t *rtcp_timestamp) const +// TODO(pbos): Make this fail when we haven't received NTP. +bool RTCPReceiver::NTP(uint32_t* ReceivedNTPsecs, + uint32_t* ReceivedNTPfrac, + uint32_t* RTCPArrivalTimeSecs, + uint32_t* RTCPArrivalTimeFrac, + uint32_t* rtcp_timestamp) const { CriticalSectionScoped lock(_criticalSectionRTCPReceiver); if(ReceivedNTPsecs) @@ -251,7 +251,7 @@ RTCPReceiver::NTP(uint32_t *ReceivedNTPsecs, if (rtcp_timestamp) { *rtcp_timestamp = _remoteSenderInfo.RTPtimeStamp; } - return 0; + return true; } bool RTCPReceiver::LastReceivedXrReferenceTimeInfo( diff --git a/modules/rtp_rtcp/source/rtcp_receiver.h b/modules/rtp_rtcp/source/rtcp_receiver.h index ebffb7cf..84eb24c7 100644 --- a/modules/rtp_rtcp/source/rtcp_receiver.h +++ b/modules/rtp_rtcp/source/rtcp_receiver.h @@ -63,11 +63,11 @@ public: char cName[RTCP_CNAME_SIZE]) const; // get received NTP - int32_t NTP(uint32_t *ReceivedNTPsecs, - uint32_t *ReceivedNTPfrac, - uint32_t *RTCPArrivalTimeSecs, - uint32_t *RTCPArrivalTimeFrac, - uint32_t *rtcp_timestamp) const; + bool NTP(uint32_t* ReceivedNTPsecs, + uint32_t* ReceivedNTPfrac, + uint32_t* RTCPArrivalTimeSecs, + uint32_t* RTCPArrivalTimeFrac, + uint32_t* rtcp_timestamp) const; bool LastReceivedXrReferenceTimeInfo(RtcpReceiveTimeInfo* info) const; diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc index 2cf7e1cb..1edbee43 100644 --- a/modules/rtp_rtcp/source/rtcp_sender.cc +++ b/modules/rtp_rtcp/source/rtcp_sender.cc @@ -65,30 +65,11 @@ std::string NACKStringBuilder::GetResult() return _stream.str(); } -RTCPSender::FeedbackState::FeedbackState(ModuleRtpRtcpImpl* module) - : send_payload_type(module->SendPayloadType()), - frequency_hz(module->CurrentSendFrequencyHz()), - packet_count_sent(module->PacketCountSent()), - byte_count_sent(module->ByteCountSent()), - module(module) { - uint32_t last_ntp_secs = 0, last_ntp_frac = 0, last_remote_sr = 0; - module->LastReceivedNTP(last_ntp_secs, last_ntp_frac, last_remote_sr); - last_rr_ntp_secs = last_ntp_secs; - last_rr_ntp_frac = last_ntp_frac; - remote_sr = last_remote_sr; - - has_last_xr_rr = module->LastReceivedXrReferenceTimeInfo(&last_xr_rr); - - uint32_t send_bitrate = 0, tmp; - module->BitrateSent(&send_bitrate, &tmp, &tmp, &tmp); - this->send_bitrate = send_bitrate; -} - RTCPSender::FeedbackState::FeedbackState() : send_payload_type(0), frequency_hz(0), - packet_count_sent(0), - byte_count_sent(0), + packets_sent(0), + media_bytes_sent(0), send_bitrate(0), last_rr_ntp_secs(0), last_rr_ntp_frac(0), @@ -654,12 +635,12 @@ int32_t RTCPSender::BuildSR(const FeedbackState& feedback_state, //sender's packet count RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, - feedback_state.packet_count_sent); + feedback_state.packets_sent); pos += 4; //sender's octet count RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, - feedback_state.byte_count_sent); + feedback_state.media_bytes_sent); pos += 4; uint8_t numberOfReportBlocks = 0; diff --git a/modules/rtp_rtcp/source/rtcp_sender.h b/modules/rtp_rtcp/source/rtcp_sender.h index fad3b5e3..26c44b04 100644 --- a/modules/rtp_rtcp/source/rtcp_sender.h +++ b/modules/rtp_rtcp/source/rtcp_sender.h @@ -51,13 +51,12 @@ class RTCPSender { public: struct FeedbackState { - explicit FeedbackState(ModuleRtpRtcpImpl* module); FeedbackState(); uint8_t send_payload_type; uint32_t frequency_hz; - uint32_t packet_count_sent; - uint32_t byte_count_sent; + uint32_t packets_sent; + uint32_t media_bytes_sent; uint32_t send_bitrate; uint32_t last_rr_ntp_secs; diff --git a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc index cba1c346..b8d53953 100644 --- a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc +++ b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc @@ -337,7 +337,7 @@ class RtcpSenderTest : public ::testing::Test { TEST_F(RtcpSenderTest, RtcpOff) { EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpOff)); - RTCPSender::FeedbackState feedback_state(rtp_rtcp_impl_); + RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState(); EXPECT_EQ(-1, rtcp_sender_->SendRTCP(feedback_state, kRtcpSr)); } @@ -381,7 +381,7 @@ TEST_F(RtcpSenderTest, TestCompound) { EXPECT_EQ(0, rtcp_sender_->SetIJStatus(true)); EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpCompound)); - RTCPSender::FeedbackState feedback_state(rtp_rtcp_impl_); + RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState(); EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpRr)); // Transmission time offset packet should be received. @@ -392,7 +392,7 @@ TEST_F(RtcpSenderTest, TestCompound) { TEST_F(RtcpSenderTest, TestCompound_NoRtpReceived) { EXPECT_EQ(0, rtcp_sender_->SetIJStatus(true)); EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpCompound)); - RTCPSender::FeedbackState feedback_state(rtp_rtcp_impl_); + RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState(); EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpRr)); // Transmission time offset packet should not be received. @@ -402,7 +402,7 @@ TEST_F(RtcpSenderTest, TestCompound_NoRtpReceived) { TEST_F(RtcpSenderTest, TestXrReceiverReferenceTime) { EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpCompound)); - RTCPSender::FeedbackState feedback_state(rtp_rtcp_impl_); + RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState(); EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state, false)); rtcp_sender_->SendRtcpXrReceiverReferenceTime(true); EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpReport)); @@ -413,7 +413,7 @@ TEST_F(RtcpSenderTest, TestXrReceiverReferenceTime) { TEST_F(RtcpSenderTest, TestNoXrReceiverReferenceTimeIfSending) { EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpCompound)); - RTCPSender::FeedbackState feedback_state(rtp_rtcp_impl_); + RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState(); EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state, true)); rtcp_sender_->SendRtcpXrReceiverReferenceTime(true); EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpReport)); @@ -424,7 +424,7 @@ TEST_F(RtcpSenderTest, TestNoXrReceiverReferenceTimeIfSending) { TEST_F(RtcpSenderTest, TestNoXrReceiverReferenceTimeIfNotEnabled) { EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpCompound)); - RTCPSender::FeedbackState feedback_state(rtp_rtcp_impl_); + RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState(); EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state, false)); rtcp_sender_->SendRtcpXrReceiverReferenceTime(false); EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpReport)); @@ -435,7 +435,7 @@ TEST_F(RtcpSenderTest, TestNoXrReceiverReferenceTimeIfNotEnabled) { TEST_F(RtcpSenderTest, TestSendTimeOfXrRrReport) { EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpCompound)); - RTCPSender::FeedbackState feedback_state(rtp_rtcp_impl_); + RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState(); EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state, false)); rtcp_sender_->SendRtcpXrReceiverReferenceTime(true); uint32_t ntp_sec; @@ -475,7 +475,7 @@ TEST_F(RtcpSenderTest, SendsTmmbnIfSetAndEmpty) { TMMBRSet bounding_set; EXPECT_EQ(0, rtcp_sender_->SetTMMBN(&bounding_set, 3)); ASSERT_EQ(0U, test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags); - RTCPSender::FeedbackState feedback_state(rtp_rtcp_impl_); + RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState(); EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state,kRtcpSr)); // We now expect the packet to show up in the rtcp_packet_info_ of // test_transport_. @@ -498,7 +498,7 @@ TEST_F(RtcpSenderTest, SendsTmmbnIfSetAndValid) { EXPECT_EQ(0, rtcp_sender_->SetTMMBN(&bounding_set, 3)); ASSERT_EQ(0U, test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags); - RTCPSender::FeedbackState feedback_state(rtp_rtcp_impl_); + RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState(); EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpSr)); // We now expect the packet to show up in the rtcp_packet_info_ of // test_transport_. diff --git a/modules/rtp_rtcp/source/rtp_header_extension.cc b/modules/rtp_rtcp/source/rtp_header_extension.cc index 2e72d750..9a1836e1 100644 --- a/modules/rtp_rtcp/source/rtp_header_extension.cc +++ b/modules/rtp_rtcp/source/rtp_header_extension.cc @@ -65,6 +65,16 @@ int32_t RtpHeaderExtensionMap::Deregister(const RTPExtensionType type) { return 0; } +bool RtpHeaderExtensionMap::IsRegistered(RTPExtensionType type) const { + std::map<uint8_t, HeaderExtension*>::const_iterator it = + extensionMap_.begin(); + for (; it != extensionMap_.end(); ++it) { + if (it->second->type == type) + return true; + } + return false; +} + int32_t RtpHeaderExtensionMap::GetType(const uint8_t id, RTPExtensionType* type) const { assert(type); diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc index 349340f5..1a3b79cb 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc +++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc @@ -230,8 +230,7 @@ int32_t ModuleRtpRtcpImpl::Process() { } if (rtcp_sender_.TimeToSendRTCPReport()) { - RTCPSender::FeedbackState feedback_state(this); - rtcp_sender_.SendRTCP(feedback_state, kRtcpReport); + rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport); } } @@ -418,12 +417,29 @@ int32_t ModuleRtpRtcpImpl::SetCSRCs( return 0; // TODO(pwestin): change to void. } -uint32_t ModuleRtpRtcpImpl::PacketCountSent() const { - return rtp_sender_.Packets(); -} +// TODO(pbos): Handle media and RTX streams separately (separate RTCP +// feedbacks). +RTCPSender::FeedbackState ModuleRtpRtcpImpl::GetFeedbackState() { + StreamDataCounters rtp_stats; + StreamDataCounters rtx_stats; + rtp_sender_.GetDataCounters(&rtp_stats, &rtx_stats); + + RTCPSender::FeedbackState state; + state.send_payload_type = SendPayloadType(); + state.frequency_hz = CurrentSendFrequencyHz(); + state.packets_sent = rtp_stats.packets + rtx_stats.packets; + state.media_bytes_sent = rtp_stats.bytes + rtx_stats.bytes; + state.module = this; + + LastReceivedNTP(&state.last_rr_ntp_secs, + &state.last_rr_ntp_frac, + &state.remote_sr); -uint32_t ModuleRtpRtcpImpl::ByteCountSent() const { - return rtp_sender_.Bytes(); + state.has_last_xr_rr = LastReceivedXrReferenceTimeInfo(&state.last_xr_rr); + + uint32_t tmp; + BitrateSent(&state.send_bitrate, &tmp, &tmp, &tmp); + return state; } int ModuleRtpRtcpImpl::CurrentSendFrequencyHz() const { @@ -433,8 +449,7 @@ int ModuleRtpRtcpImpl::CurrentSendFrequencyHz() const { int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) { if (rtcp_sender_.Sending() != sending) { // Sends RTCP BYE when going from true to false - RTCPSender::FeedbackState feedback_state(this); - if (rtcp_sender_.SetSendingStatus(feedback_state, sending) != 0) { + if (rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending) != 0) { LOG(LS_WARNING) << "Failed to send RTCP BYE"; } @@ -499,8 +514,7 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData( if (!IsDefaultModule()) { // Don't send RTCP from default module. if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) { - RTCPSender::FeedbackState feedback_state(this); - rtcp_sender_.SendRTCP(feedback_state, kRtcpReport); + rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport); } return rtp_sender_.SendOutgoingData(frame_type, payload_type, @@ -599,14 +613,10 @@ int ModuleRtpRtcpImpl::TimeToSendPadding(int bytes) { } } else { CriticalSectionScoped lock(critical_section_module_ptrs_.get()); - // Decide what media stream to pad on based on a round-robin scheme. for (size_t i = 0; i < child_modules_.size(); ++i) { - padding_index_ = (padding_index_ + 1) % child_modules_.size(); // Send padding on one of the modules sending media. - if (child_modules_[padding_index_]->SendingMedia() && - child_modules_[padding_index_]->rtp_sender_.GetTargetBitrate() > 0) { - return child_modules_[padding_index_]->rtp_sender_.TimeToSendPadding( - bytes); + if (child_modules_[i]->SendingMedia()) { + return child_modules_[i]->rtp_sender_.TimeToSendPadding(bytes); } } } @@ -751,7 +761,9 @@ int32_t ModuleRtpRtcpImpl::RemoteNTP( received_ntpfrac, rtcp_arrival_time_secs, rtcp_arrival_time_frac, - rtcp_timestamp); + rtcp_timestamp) + ? 0 + : -1; } // Get RoundTripTime. @@ -782,8 +794,7 @@ int32_t ModuleRtpRtcpImpl::ResetSendDataCountersRTP() { // Force a send of an RTCP packet. // Normal SR and RR are triggered via the process function. int32_t ModuleRtpRtcpImpl::SendRTCP(uint32_t rtcp_packet_type) { - RTCPSender::FeedbackState feedback_state(this); - return rtcp_sender_.SendRTCP(feedback_state, rtcp_packet_type); + return rtcp_sender_.SendRTCP(GetFeedbackState(), rtcp_packet_type); } int32_t ModuleRtpRtcpImpl::SetRTCPApplicationSpecificData( @@ -811,11 +822,17 @@ bool ModuleRtpRtcpImpl::RtcpXrRrtrStatus() const { int32_t ModuleRtpRtcpImpl::DataCountersRTP( uint32_t* bytes_sent, uint32_t* packets_sent) const { + StreamDataCounters rtp_stats; + StreamDataCounters rtx_stats; + rtp_sender_.GetDataCounters(&rtp_stats, &rtx_stats); + if (bytes_sent) { - *bytes_sent = rtp_sender_.Bytes(); + *bytes_sent = rtp_stats.bytes + rtp_stats.padding_bytes + + rtp_stats.header_bytes + rtx_stats.bytes + + rtx_stats.padding_bytes + rtx_stats.header_bytes; } if (packets_sent) { - *packets_sent = rtp_sender_.Packets(); + *packets_sent = rtp_stats.packets + rtx_stats.packets; } return 0; } @@ -955,9 +972,8 @@ int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list, } nack_last_seq_number_sent_ = nack_list[start_id + nackLength - 1]; - RTCPSender::FeedbackState feedback_state(this); return rtcp_sender_.SendRTCP( - feedback_state, kRtcpNack, nackLength, &nack_list[start_id]); + GetFeedbackState(), kRtcpNack, nackLength, &nack_list[start_id]); } // Store the sent packets, needed to answer to a Negative acknowledgment @@ -1074,9 +1090,8 @@ int32_t ModuleRtpRtcpImpl::RequestKeyFrame() { int32_t ModuleRtpRtcpImpl::SendRTCPSliceLossIndication( const uint8_t picture_id) { - RTCPSender::FeedbackState feedback_state(this); return rtcp_sender_.SendRTCP( - feedback_state, kRtcpSli, 0, 0, false, picture_id); + GetFeedbackState(), kRtcpSli, 0, 0, false, picture_id); } int32_t ModuleRtpRtcpImpl::SetCameraDelay(const int32_t delay_ms) { @@ -1245,9 +1260,8 @@ void ModuleRtpRtcpImpl::OnRequestSendReport() { int32_t ModuleRtpRtcpImpl::SendRTCPReferencePictureSelection( const uint64_t picture_id) { - RTCPSender::FeedbackState feedback_state(this); return rtcp_sender_.SendRTCP( - feedback_state, kRtcpRpsi, 0, 0, false, picture_id); + GetFeedbackState(), kRtcpRpsi, 0, 0, false, picture_id); } uint32_t ModuleRtpRtcpImpl::SendTimeOfSendReport( @@ -1274,23 +1288,24 @@ void ModuleRtpRtcpImpl::OnReceivedNACK( rtp_sender_.OnReceivedNACK(nack_sequence_numbers, rtt); } -int32_t ModuleRtpRtcpImpl::LastReceivedNTP( - uint32_t& rtcp_arrival_time_secs, // When we got the last report. - uint32_t& rtcp_arrival_time_frac, - uint32_t& remote_sr) { +bool ModuleRtpRtcpImpl::LastReceivedNTP( + uint32_t* rtcp_arrival_time_secs, // When we got the last report. + uint32_t* rtcp_arrival_time_frac, + uint32_t* remote_sr) const { // Remote SR: NTP inside the last received (mid 16 bits from sec and frac). uint32_t ntp_secs = 0; uint32_t ntp_frac = 0; - if (-1 == rtcp_receiver_.NTP(&ntp_secs, - &ntp_frac, - &rtcp_arrival_time_secs, - &rtcp_arrival_time_frac, - NULL)) { - return -1; + if (!rtcp_receiver_.NTP(&ntp_secs, + &ntp_frac, + rtcp_arrival_time_secs, + rtcp_arrival_time_frac, + NULL)) { + return false; } - remote_sr = ((ntp_secs & 0x0000ffff) << 16) + ((ntp_frac & 0xffff0000) >> 16); - return 0; + *remote_sr = + ((ntp_secs & 0x0000ffff) << 16) + ((ntp_frac & 0xffff0000) >> 16); + return true; } bool ModuleRtpRtcpImpl::LastReceivedXrReferenceTimeInfo( diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h index 7e7ea027..4a23dd40 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h +++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h @@ -89,12 +89,10 @@ class ModuleRtpRtcpImpl : public RtpRtcp { virtual int32_t SetCSRCStatus(const bool include) OVERRIDE; - virtual uint32_t PacketCountSent() const; + virtual RTCPSender::FeedbackState GetFeedbackState(); virtual int CurrentSendFrequencyHz() const; - virtual uint32_t ByteCountSent() const; - virtual void SetRTXSendStatus(const int mode) OVERRIDE; virtual void RTXSendStatus(int* mode, uint32_t* ssrc, @@ -328,9 +326,9 @@ class ModuleRtpRtcpImpl : public RtpRtcp { const FecProtectionParams* delta_params, const FecProtectionParams* key_params) OVERRIDE; - virtual int32_t LastReceivedNTP(uint32_t& NTPsecs, - uint32_t& NTPfrac, - uint32_t& remote_sr); + virtual bool LastReceivedNTP(uint32_t* NTPsecs, + uint32_t* NTPfrac, + uint32_t* remote_sr) const; virtual bool LastReceivedXrReferenceTimeInfo(RtcpReceiveTimeInfo* info) const; diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc index 930778c3..3101a1fe 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc @@ -388,7 +388,7 @@ class RtpSendingTest : public ::testing::Test { VideoCodec codec_; }; -TEST_F(RtpSendingTest, RoundRobinPadding) { +TEST_F(RtpSendingTest, DISABLED_RoundRobinPadding) { // We have to send on an SSRC to be allowed to pad, since a marker bit must // be sent prior to padding packets. const uint8_t payload[200] = {0}; @@ -410,9 +410,13 @@ TEST_F(RtpSendingTest, RoundRobinPadding) { ExpectPadding(expected_padding); } -TEST_F(RtpSendingTest, RoundRobinPaddingRtx) { +TEST_F(RtpSendingTest, DISABLED_RoundRobinPaddingRtx) { // Enable RTX to allow padding to be sent prior to media. for (int i = 1; i < codec_.numberOfSimulcastStreams + 1; ++i) { + // Abs-send-time is needed to be allowed to send padding prior to media, + // as otherwise the timestmap used for BWE will be broken. + senders_[i]->RegisterSendRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, + 1); senders_[i]->SetRtxSendPayloadType(96); senders_[i]->SetRtxSsrc(kSenderRtxSsrc + i); senders_[i]->SetRTXSendStatus(kRtxRetransmitted); @@ -436,7 +440,7 @@ TEST_F(RtpSendingTest, RoundRobinPaddingRtx) { ExpectPadding(expected_padding); } -TEST_F(RtpSendingTest, RoundRobinPaddingRtxRedundantPayloads) { +TEST_F(RtpSendingTest, DISABLED_RoundRobinPaddingRtxRedundantPayloads) { for (int i = 1; i < codec_.numberOfSimulcastStreams + 1; ++i) { senders_[i]->SetRtxSendPayloadType(96); senders_[i]->SetRtxSsrc(kSenderRtxSsrc + i); diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc index c24b15a3..eb88be7c 100644 --- a/modules/rtp_rtcp/source/rtp_sender.cc +++ b/modules/rtp_rtcp/source/rtp_sender.cc @@ -87,6 +87,7 @@ RTPSender::RTPSender(const int32_t id, timestamp_(0), capture_time_ms_(0), last_timestamp_time_ms_(0), + media_has_been_sent_(false), last_packet_marker_bit_(false), num_csrcs_(0), csrcs_(), @@ -430,14 +431,9 @@ int32_t RTPSender::SendOutgoingData( "Send", "type", FrameTypeToString(frame_type)); assert(frame_type != kAudioFrameSpeech && frame_type != kAudioFrameCN); - if (frame_type == kFrameEmpty) { - if (paced_sender_->Enabled()) { - // Padding is driven by the pacer and not by the encoder. - return 0; - } - return SendPaddingAccordingToBitrate(payload_type, capture_timestamp, - capture_time_ms) ? 0 : -1; - } + if (frame_type == kFrameEmpty) + return 0; + ret_val = video_->SendVideo(video_type, frame_type, payload_type, capture_timestamp, capture_time_ms, payload_data, payload_size, @@ -475,45 +471,6 @@ int RTPSender::SendRedundantPayloads(int payload_type, int bytes_to_send) { return bytes_to_send - bytes_left; } -bool RTPSender::SendPaddingAccordingToBitrate( - int8_t payload_type, uint32_t capture_timestamp, - int64_t capture_time_ms) { - // Current bitrate since last estimate(1 second) averaged with the - // estimate since then, to get the most up to date bitrate. - uint32_t current_bitrate = bitrate_sent_.BitrateNow(); - uint32_t target_bitrate = GetTargetBitrate(); - int bitrate_diff = target_bitrate - current_bitrate; - if (bitrate_diff <= 0) { - return true; - } - int bytes = 0; - if (current_bitrate == 0) { - // Start up phase. Send one 33.3 ms batch to start with. - bytes = (bitrate_diff / 8) / 30; - } else { - bytes = (bitrate_diff / 8); - // Cap at 200 ms of target send data. - int bytes_cap = target_bitrate / 1000 * 25; // 1000 / 8 / 5. - if (bytes > bytes_cap) { - bytes = bytes_cap; - } - } - uint32_t timestamp; - { - CriticalSectionScoped cs(send_critsect_); - // Add the random RTP timestamp offset and store the capture time for - // later calculation of the send time offset. - timestamp = start_timestamp_ + capture_timestamp; - timestamp_ = timestamp; - capture_time_ms_ = capture_time_ms; - last_timestamp_time_ms_ = clock_->TimeInMilliseconds(); - } - int bytes_sent = SendPadData(payload_type, timestamp, capture_time_ms, - bytes, false, false); - // We did not manage to send all bytes. Comparing with 31 due to modulus 32. - return bytes - bytes_sent < 31; -} - int RTPSender::BuildPaddingPacket(uint8_t* packet, int header_length, int32_t bytes) { int padding_bytes_in_packet = kMaxPaddingLength; @@ -536,9 +493,7 @@ int RTPSender::BuildPaddingPacket(uint8_t* packet, int header_length, int RTPSender::SendPadData(int payload_type, uint32_t timestamp, int64_t capture_time_ms, - int32_t bytes, - bool force_full_size_packets, - bool over_rtx) { + int32_t bytes) { // Drop this packet if we're not sending media packets. if (!SendingMedia()) { return bytes; @@ -547,36 +502,34 @@ int RTPSender::SendPadData(int payload_type, int bytes_sent = 0; for (; bytes > 0; bytes -= padding_bytes_in_packet) { // Always send full padding packets. - if (force_full_size_packets && bytes < kMaxPaddingLength) + if (bytes < kMaxPaddingLength) bytes = kMaxPaddingLength; - if (bytes < kMaxPaddingLength) { - if (force_full_size_packets) { - bytes = kMaxPaddingLength; - } else { - // Round to the nearest multiple of 32. - bytes = (bytes + 16) & 0xffe0; - } - } - if (bytes < 32) { - // Sanity don't send empty packets. - break; - } + uint32_t ssrc; uint16_t sequence_number; + bool over_rtx; { CriticalSectionScoped cs(send_critsect_); // Only send padding packets following the last packet of a frame, // indicated by the marker bit. - if (!over_rtx && !last_packet_marker_bit_) - return bytes_sent; if (rtx_ == kRtxOff) { + // Without RTX we can't send padding in the middle of frames. + if (!last_packet_marker_bit_) + return bytes_sent; ssrc = ssrc_; sequence_number = sequence_number_; ++sequence_number_; + over_rtx = false; } else { + // Without abs-send-time a media packet must be sent before padding so + // that the timestamps used for estimation are correct. + if (!media_has_been_sent_ && !rtp_header_extension_map_.IsRegistered( + kRtpExtensionAbsoluteSendTime)) + return bytes_sent; ssrc = ssrc_rtx_; sequence_number = sequence_number_rtx_; ++sequence_number_rtx_; + over_rtx = true; } } @@ -656,10 +609,13 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) { return length; } } - - CriticalSectionScoped lock(send_critsect_); + int rtx = kRtxOff; + { + CriticalSectionScoped lock(send_critsect_); + rtx = rtx_; + } return PrepareAndSendPacket(data_buffer, length, capture_time_ms, - (rtx_ & kRtxRetransmitted) > 0, true) ? + (rtx & kRtxRetransmitted) > 0, true) ? length : -1; } @@ -852,6 +808,10 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer, diff_ms); UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms); bool ret = SendPacketToNetwork(buffer_to_send_ptr, length); + if (ret) { + CriticalSectionScoped lock(send_critsect_); + media_has_been_sent_ = true; + } UpdateRtpStats(buffer_to_send_ptr, length, rtp_header, send_over_rtx, is_retransmit); return ret; @@ -907,6 +867,7 @@ bool RTPSender::IsFecPacket(const uint8_t* buffer, } int RTPSender::TimeToSendPadding(int bytes) { + assert(bytes > 0); int payload_type; int64_t capture_time_ms; uint32_t timestamp; @@ -933,12 +894,8 @@ int RTPSender::TimeToSendPadding(int bytes) { bytes_sent = SendRedundantPayloads(payload_type, bytes); bytes -= bytes_sent; if (bytes > 0) { - int padding_sent = SendPadData(payload_type, - timestamp, - capture_time_ms, - bytes, - true, - rtx != kRtxOff); + int padding_sent = + SendPadData(payload_type, timestamp, capture_time_ms, bytes); bytes_sent += padding_sent; } return bytes_sent; @@ -992,6 +949,11 @@ int32_t RTPSender::SendToNetwork( uint32_t length = payload_length + rtp_header_length; if (!SendPacketToNetwork(buffer, length)) return -1; + assert(payload_length - rtp_header.paddingLength > 0); + { + CriticalSectionScoped lock(send_critsect_); + media_has_been_sent_ = true; + } UpdateRtpStats(buffer, length, rtp_header, false, false); return 0; } @@ -1061,17 +1023,11 @@ void RTPSender::ResetDataCounters() { } } -uint32_t RTPSender::Packets() const { - CriticalSectionScoped lock(statistics_crit_.get()); - return rtp_stats_.packets + rtx_rtp_stats_.packets; -} - -// Number of sent RTP bytes. -uint32_t RTPSender::Bytes() const { +void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats, + StreamDataCounters* rtx_stats) const { CriticalSectionScoped lock(statistics_crit_.get()); - return rtp_stats_.bytes + rtp_stats_.header_bytes + rtp_stats_.padding_bytes + - rtx_rtp_stats_.bytes + rtx_rtp_stats_.header_bytes + - rtx_rtp_stats_.padding_bytes; + *rtp_stats = rtp_stats_; + *rtx_stats = rtx_rtp_stats_; } int RTPSender::CreateRTPHeader( @@ -1732,6 +1688,7 @@ void RTPSender::SetRtpState(const RtpState& rtp_state) { timestamp_ = rtp_state.timestamp; capture_time_ms_ = rtp_state.capture_time_ms; last_timestamp_time_ms_ = rtp_state.last_timestamp_time_ms; + media_has_been_sent_ = rtp_state.media_has_been_sent; } RtpState RTPSender::GetRtpState() const { @@ -1743,6 +1700,7 @@ RtpState RTPSender::GetRtpState() const { state.timestamp = timestamp_; state.capture_time_ms = capture_time_ms_; state.last_timestamp_time_ms = last_timestamp_time_ms_; + state.media_has_been_sent = media_has_been_sent_; return state; } diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h index 4a9e10ed..f3a2bdc1 100644 --- a/modules/rtp_rtcp/source/rtp_sender.h +++ b/modules/rtp_rtcp/source/rtp_sender.h @@ -109,11 +109,8 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { void SetSendingMediaStatus(const bool enabled); bool SendingMedia() const; - // Number of sent RTP packets. - uint32_t Packets() const; - - // Number of sent RTP bytes. - uint32_t Bytes() const; + void GetDataCounters(StreamDataCounters* rtp_stats, + StreamDataCounters* rtx_stats) const; void ResetDataCounters(); @@ -270,9 +267,7 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { int SendPadData(int payload_type, uint32_t timestamp, int64_t capture_time_ms, - int32_t bytes, - bool force_full_size_packets, - bool only_pad_after_markerbit); + int32_t bytes); // Called on update of RTP statistics. void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback); @@ -312,9 +307,6 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { int SendRedundantPayloads(int payload_type, int bytes); - bool SendPaddingAccordingToBitrate(int8_t payload_type, - uint32_t capture_timestamp, - int64_t capture_time_ms); int BuildPaddingPacket(uint8_t* packet, int header_length, int32_t bytes); void BuildRtxPacket(uint8_t* buffer, uint16_t* length, @@ -395,6 +387,7 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { uint32_t timestamp_ GUARDED_BY(send_critsect_); int64_t capture_time_ms_ GUARDED_BY(send_critsect_); int64_t last_timestamp_time_ms_ GUARDED_BY(send_critsect_); + bool media_has_been_sent_ GUARDED_BY(send_critsect_); bool last_packet_marker_bit_ GUARDED_BY(send_critsect_); uint8_t num_csrcs_ GUARDED_BY(send_critsect_); uint32_t csrcs_[kRtpCsrcSize] GUARDED_BY(send_critsect_); diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc index 40b10548..e9b01def 100644 --- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc @@ -39,6 +39,7 @@ const uint8_t kAudioLevel = 0x5a; const uint8_t kAudioLevelExtensionId = 9; const int kAudioPayload = 103; const uint64_t kStartTime = 123456789; +const size_t kMaxPaddingSize = 224u; } // namespace using testing::_; @@ -700,7 +701,7 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) { kAbsoluteSendTimeExtensionId); rtp_sender_->SetTargetBitrate(300000); const size_t kNumPayloadSizes = 10; - const int kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700, 750, + const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700, 750, 800, 850, 900, 950}; // Send 10 packets of increasing size. for (size_t i = 0; i < kNumPayloadSizes; ++i) { @@ -711,25 +712,27 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) { rtp_sender_->TimeToSendPacket(seq_num++, capture_time_ms, false); fake_clock_.AdvanceTimeMilliseconds(33); } - const int kPaddingPayloadSize = 224; // The amount of padding to send it too small to send a payload packet. - EXPECT_CALL(transport, SendPacket(_, _, kPaddingPayloadSize + rtp_header_len)) + EXPECT_CALL(transport, + SendPacket(_, _, kMaxPaddingSize + rtp_header_len)) .WillOnce(testing::ReturnArg<2>()); - EXPECT_EQ(kPaddingPayloadSize, rtp_sender_->TimeToSendPadding(49)); + EXPECT_EQ(kMaxPaddingSize, + static_cast<size_t>(rtp_sender_->TimeToSendPadding(49))); const int kRtxHeaderSize = 2; EXPECT_CALL(transport, SendPacket(_, _, kPayloadSizes[0] + rtp_header_len + kRtxHeaderSize)) .WillOnce(testing::ReturnArg<2>()); - EXPECT_EQ(kPayloadSizes[0], rtp_sender_->TimeToSendPadding(500)); + EXPECT_EQ(kPayloadSizes[0], + static_cast<size_t>(rtp_sender_->TimeToSendPadding(500))); EXPECT_CALL(transport, SendPacket(_, _, kPayloadSizes[kNumPayloadSizes - 1] + rtp_header_len + kRtxHeaderSize)) .WillOnce(testing::ReturnArg<2>()); - EXPECT_CALL(transport, SendPacket(_, _, kPaddingPayloadSize + rtp_header_len)) + EXPECT_CALL(transport, SendPacket(_, _, kMaxPaddingSize + rtp_header_len)) .WillOnce(testing::ReturnArg<2>()); - EXPECT_EQ(kPayloadSizes[kNumPayloadSizes - 1] + kPaddingPayloadSize, - rtp_sender_->TimeToSendPadding(999)); + EXPECT_EQ(kPayloadSizes[kNumPayloadSizes - 1] + kMaxPaddingSize, + static_cast<size_t>(rtp_sender_->TimeToSendPadding(999))); } TEST_F(RtpSenderTest, SendGenericVideo) { @@ -959,7 +962,6 @@ TEST_F(RtpSenderTest, StreamDataCountersCallbacks) { const uint8_t kRedPayloadType = 96; const uint8_t kUlpfecPayloadType = 97; - const uint32_t kMaxPaddingSize = 224; char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; const uint8_t payload_type = 127; ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, @@ -988,7 +990,7 @@ TEST_F(RtpSenderTest, StreamDataCountersCallbacks) { // Send padding. rtp_sender_->TimeToSendPadding(kMaxPaddingSize); // {bytes = 6, header = 24, padding = 224, packets = 3, retrans = 1, fec = 0} - EXPECT_TRUE(callback.Matches(ssrc, 6, 24, 224, 3, 1, 0)); + EXPECT_TRUE(callback.Matches(ssrc, 6, 24, kMaxPaddingSize, 3, 1, 0)); // Send FEC. rtp_sender_->SetGenericFECStatus(true, kRedPayloadType, kUlpfecPayloadType); @@ -1003,7 +1005,7 @@ TEST_F(RtpSenderTest, StreamDataCountersCallbacks) { sizeof(payload), NULL)); // {bytes = 34, header = 48, padding = 224, packets = 5, retrans = 1, fec = 1} - EXPECT_TRUE(callback.Matches(ssrc, 34, 48, 224, 5, 1, 1)); + EXPECT_TRUE(callback.Matches(ssrc, 34, 48, kMaxPaddingSize, 5, 1, 1)); rtp_sender_->RegisterRtpStatisticsCallback(NULL); } @@ -1093,13 +1095,25 @@ TEST_F(RtpSenderTest, BytesReportedCorrectly) { sizeof(payload), 0)); - EXPECT_GT(transport_.total_bytes_sent_, 0u); - EXPECT_EQ(transport_.total_bytes_sent_, rtp_sender_->Bytes()); - size_t last_bytes_sent = transport_.total_bytes_sent_; - - rtp_sender_->TimeToSendPadding(42); - - EXPECT_GT(transport_.total_bytes_sent_, last_bytes_sent); - EXPECT_EQ(transport_.total_bytes_sent_, rtp_sender_->Bytes()); + // Will send 2 full-size padding packets. + rtp_sender_->TimeToSendPadding(1); + rtp_sender_->TimeToSendPadding(1); + + StreamDataCounters rtp_stats; + StreamDataCounters rtx_stats; + rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats); + + // Payload + 1-byte generic header. + EXPECT_EQ(rtp_stats.bytes, sizeof(payload) + 1); + EXPECT_EQ(rtp_stats.header_bytes, 12u); + EXPECT_EQ(rtp_stats.padding_bytes, 0u); + EXPECT_EQ(rtx_stats.bytes, 0u); + EXPECT_EQ(rtx_stats.header_bytes, 24u); + EXPECT_EQ(rtx_stats.padding_bytes, 2 * kMaxPaddingSize); + + EXPECT_EQ(transport_.total_bytes_sent_, + rtp_stats.bytes + rtp_stats.header_bytes + rtp_stats.padding_bytes + + rtx_stats.bytes + rtx_stats.header_bytes + + rtx_stats.padding_bytes); } } // namespace webrtc diff --git a/modules/rtp_rtcp/source/ssrc_database.cc b/modules/rtp_rtcp/source/ssrc_database.cc index df09b01b..4e230833 100644 --- a/modules/rtp_rtcp/source/ssrc_database.cc +++ b/modules/rtp_rtcp/source/ssrc_database.cc @@ -57,45 +57,12 @@ SSRCDatabase::CreateSSRC() uint32_t ssrc = GenerateRandom(); -#ifndef WEBRTC_NO_STL - while(_ssrcMap.find(ssrc) != _ssrcMap.end()) { ssrc = GenerateRandom(); } _ssrcMap[ssrc] = 0; -#else - if(_sizeOfSSRC <= _numberOfSSRC) - { - // allocate more space - const int newSize = _sizeOfSSRC + 10; - uint32_t* tempSSRCVector = new uint32_t[newSize]; - memcpy(tempSSRCVector, _ssrcVector, _sizeOfSSRC*sizeof(uint32_t)); - delete [] _ssrcVector; - - _ssrcVector = tempSSRCVector; - _sizeOfSSRC = newSize; - } - - // check if in DB - if(_ssrcVector) - { - for (int i=0; i<_numberOfSSRC; i++) - { - if (_ssrcVector[i] == ssrc) - { - // we have a match - i = 0; // start over with a new ssrc - ssrc = GenerateRandom(); - } - - } - // add to database - _ssrcVector[_numberOfSSRC] = ssrc; - _numberOfSSRC++; - } -#endif return ssrc; } @@ -103,39 +70,7 @@ int32_t SSRCDatabase::RegisterSSRC(const uint32_t ssrc) { CriticalSectionScoped lock(_critSect); - -#ifndef WEBRTC_NO_STL - _ssrcMap[ssrc] = 0; - -#else - if(_sizeOfSSRC <= _numberOfSSRC) - { - // allocate more space - const int newSize = _sizeOfSSRC + 10; - uint32_t* tempSSRCVector = new uint32_t[newSize]; - memcpy(tempSSRCVector, _ssrcVector, _sizeOfSSRC*sizeof(uint32_t)); - delete [] _ssrcVector; - - _ssrcVector = tempSSRCVector; - _sizeOfSSRC = newSize; - } - // check if in DB - if(_ssrcVector) - { - for (int i=0; i<_numberOfSSRC; i++) - { - if (_ssrcVector[i] == ssrc) - { - // we have a match - return -1; - } - } - // add to database - _ssrcVector[_numberOfSSRC] = ssrc; - _numberOfSSRC++; - } -#endif return 0; } @@ -143,26 +78,7 @@ int32_t SSRCDatabase::ReturnSSRC(const uint32_t ssrc) { CriticalSectionScoped lock(_critSect); - -#ifndef WEBRTC_NO_STL _ssrcMap.erase(ssrc); - -#else - if(_ssrcVector) - { - for (int i=0; i<_numberOfSSRC; i++) - { - if (_ssrcVector[i] == ssrc) - { - // we have a match - // remove from database - _ssrcVector[i] = _ssrcVector[_numberOfSSRC-1]; - _numberOfSSRC--; - break; - } - } - } -#endif return 0; } @@ -178,21 +94,12 @@ SSRCDatabase::SSRCDatabase() srand(tv.tv_usec); #endif -#ifdef WEBRTC_NO_STL - _sizeOfSSRC = 10; - _numberOfSSRC = 0; - _ssrcVector = new uint32_t[10]; -#endif _critSect = CriticalSectionWrapper::CreateCriticalSection(); } SSRCDatabase::~SSRCDatabase() { -#ifdef WEBRTC_NO_STL - delete [] _ssrcVector; -#else _ssrcMap.clear(); -#endif delete _critSect; } diff --git a/modules/rtp_rtcp/source/ssrc_database.h b/modules/rtp_rtcp/source/ssrc_database.h index e1f90e79..2d4932af 100644 --- a/modules/rtp_rtcp/source/ssrc_database.h +++ b/modules/rtp_rtcp/source/ssrc_database.h @@ -11,9 +11,7 @@ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_ -#ifndef WEBRTC_NO_STL #include <map> -#endif #include "webrtc/system_wrappers/interface/static_instance.h" #include "webrtc/typedefs.h" @@ -46,14 +44,7 @@ private: uint32_t GenerateRandom(); -#ifdef WEBRTC_NO_STL - int _numberOfSSRC; - int _sizeOfSSRC; - - uint32_t* _ssrcVector; -#else std::map<uint32_t, uint32_t> _ssrcMap; -#endif CriticalSectionWrapper* _critSect; }; diff --git a/modules/video_capture/ios/rtc_video_capture_ios_objc.mm b/modules/video_capture/ios/rtc_video_capture_ios_objc.mm index 641ca241..ac90b72a 100644 --- a/modules/video_capture/ios/rtc_video_capture_ios_objc.mm +++ b/modules/video_capture/ios/rtc_video_capture_ios_objc.mm @@ -43,6 +43,12 @@ using namespace webrtc::videocapturemodule; _owner = owner; _captureId = captureId; _captureSession = [[AVCaptureSession alloc] init]; +#if defined(__IPHONE_7_0) && __IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_7_0 + NSString* version = [[UIDevice currentDevice] systemVersion]; + if ([version integerValue] >= 7) { + _captureSession.usesApplicationAudioSession = NO; + } +#endif _captureChanging = NO; _captureChangingCondition = [[NSCondition alloc] init]; @@ -217,6 +223,9 @@ using namespace webrtc::videocapturemodule; return; switch ([UIApplication sharedApplication].statusBarOrientation) { case UIInterfaceOrientationPortrait: +#if defined(__IPHONE_8_0) && __IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_8_0 + case UIInterfaceOrientationUnknown: +#endif _connection.videoOrientation = AVCaptureVideoOrientationPortrait; break; case UIInterfaceOrientationPortraitUpsideDown: diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn index 0dc6721b..cbd12936 100644 --- a/modules/video_coding/BUILD.gn +++ b/modules/video_coding/BUILD.gn @@ -75,8 +75,6 @@ source_set("video_coding") { source_set("video_coding_utility") { sources = [ - "utility/exp_filter.cc", - "utility/include/exp_filter.h", "utility/include/frame_dropper.h", "utility/frame_dropper.cc", ] diff --git a/modules/video_coding/main/source/media_opt_util.cc b/modules/video_coding/main/source/media_opt_util.cc index ba86575f..b506a5b5 100644 --- a/modules/video_coding/main/source/media_opt_util.cc +++ b/modules/video_coding/main/source/media_opt_util.cc @@ -837,7 +837,7 @@ uint8_t VCMLossProtectionLogic::FilteredLoss( case kNoFilter: break; case kAvgFilter: - filtered_loss = static_cast<uint8_t> (_lossPr255.Value() + 0.5); + filtered_loss = static_cast<uint8_t>(_lossPr255.filtered() + 0.5); break; case kMaxFilter: filtered_loss = MaxFilteredLossPr(nowMs); @@ -907,8 +907,8 @@ VCMLossProtectionLogic::UpdateMethod() _currentParameters.keyFrameSize = _keyFrameSize; _currentParameters.fecRateDelta = _fecRateDelta; _currentParameters.fecRateKey = _fecRateKey; - _currentParameters.packetsPerFrame = _packetsPerFrame.Value(); - _currentParameters.packetsPerFrameKey = _packetsPerFrameKey.Value(); + _currentParameters.packetsPerFrame = _packetsPerFrame.filtered(); + _currentParameters.packetsPerFrameKey = _packetsPerFrameKey.filtered(); _currentParameters.residualPacketLossFec = _residualPacketLossFec; _currentParameters.codecWidth = _codecWidth; _currentParameters.codecHeight = _codecHeight; diff --git a/modules/video_coding/main/source/media_opt_util.h b/modules/video_coding/main/source/media_opt_util.h index f39a5780..d421d9e7 100644 --- a/modules/video_coding/main/source/media_opt_util.h +++ b/modules/video_coding/main/source/media_opt_util.h @@ -14,9 +14,9 @@ #include <math.h> #include <stdlib.h> +#include "webrtc/base/exp_filter.h" #include "webrtc/modules/video_coding/main/source/internal_defines.h" #include "webrtc/modules/video_coding/main/source/qm_select.h" -#include "webrtc/modules/video_coding/utility/include/exp_filter.h" #include "webrtc/system_wrappers/interface/trace.h" #include "webrtc/typedefs.h" @@ -367,27 +367,27 @@ private: // Sets the available loss protection methods. void UpdateMaxLossHistory(uint8_t lossPr255, int64_t now); uint8_t MaxFilteredLossPr(int64_t nowMs) const; - VCMProtectionMethod* _selectedMethod; - VCMProtectionParameters _currentParameters; - uint32_t _rtt; - float _lossPr; - float _bitRate; - float _frameRate; - float _keyFrameSize; - uint8_t _fecRateKey; - uint8_t _fecRateDelta; - int64_t _lastPrUpdateT; - int64_t _lastPacketPerFrameUpdateT; - int64_t _lastPacketPerFrameUpdateTKey; - VCMExpFilter _lossPr255; - VCMLossProbabilitySample _lossPrHistory[kLossPrHistorySize]; - uint8_t _shortMaxLossPr255; - VCMExpFilter _packetsPerFrame; - VCMExpFilter _packetsPerFrameKey; - float _residualPacketLossFec; - uint16_t _codecWidth; - uint16_t _codecHeight; - int _numLayers; + VCMProtectionMethod* _selectedMethod; + VCMProtectionParameters _currentParameters; + uint32_t _rtt; + float _lossPr; + float _bitRate; + float _frameRate; + float _keyFrameSize; + uint8_t _fecRateKey; + uint8_t _fecRateDelta; + int64_t _lastPrUpdateT; + int64_t _lastPacketPerFrameUpdateT; + int64_t _lastPacketPerFrameUpdateTKey; + rtc::ExpFilter _lossPr255; + VCMLossProbabilitySample _lossPrHistory[kLossPrHistorySize]; + uint8_t _shortMaxLossPr255; + rtc::ExpFilter _packetsPerFrame; + rtc::ExpFilter _packetsPerFrameKey; + float _residualPacketLossFec; + uint16_t _codecWidth; + uint16_t _codecHeight; + int _numLayers; }; } // namespace media_optimization diff --git a/modules/video_coding/utility/exp_filter.cc b/modules/video_coding/utility/exp_filter.cc deleted file mode 100644 index 44f280bc..00000000 --- a/modules/video_coding/utility/exp_filter.cc +++ /dev/null @@ -1,60 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/video_coding/utility/include/exp_filter.h" - -#include <math.h> - -namespace webrtc { - -void -VCMExpFilter::Reset(float alpha) -{ - _alpha = alpha; - _filtered = -1.0; -} - -float -VCMExpFilter::Apply(float exp, float sample) -{ - if (_filtered == -1.0) - { - // Initialize filtered bit rates - _filtered = sample; - } - else if (exp == 1.0) - { - _filtered = _alpha * _filtered + (1 - _alpha) * sample; - } - else - { - float alpha = pow(_alpha, exp); - _filtered = alpha * _filtered + (1 - alpha) * sample; - } - if (_max != -1 && _filtered > _max) - { - _filtered = _max; - } - return _filtered; -} - -void -VCMExpFilter::UpdateBase(float alpha) -{ - _alpha = alpha; -} - -float -VCMExpFilter::Value() const -{ - return _filtered; -} - -} diff --git a/modules/video_coding/utility/frame_dropper.cc b/modules/video_coding/utility/frame_dropper.cc index d3c25fb9..54c8cb8a 100644 --- a/modules/video_coding/utility/frame_dropper.cc +++ b/modules/video_coding/utility/frame_dropper.cc @@ -86,25 +86,27 @@ FrameDropper::Fill(uint32_t frameSizeBytes, bool deltaFrame) { _keyFrameSizeAvgKbits.Apply(1, frameSizeKbits); _keyFrameRatio.Apply(1.0, 1.0); - if (frameSizeKbits > _keyFrameSizeAvgKbits.Value()) + if (frameSizeKbits > _keyFrameSizeAvgKbits.filtered()) { // Remove the average key frame size since we // compensate for key frames when adding delta // frames. - frameSizeKbits -= _keyFrameSizeAvgKbits.Value(); + frameSizeKbits -= _keyFrameSizeAvgKbits.filtered(); } else { // Shouldn't be negative, so zero is the lower bound. frameSizeKbits = 0; } - if (_keyFrameRatio.Value() > 1e-5 && 1 / _keyFrameRatio.Value() < _keyFrameSpreadFrames) + if (_keyFrameRatio.filtered() > 1e-5 && + 1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames) { // We are sending key frames more often than our upper bound for // how much we allow the key frame compensation to be spread // out in time. Therefor we must use the key frame ratio rather // than keyFrameSpreadFrames. - _keyFrameCount = static_cast<int32_t>(1 / _keyFrameRatio.Value() + 0.5); + _keyFrameCount = + static_cast<int32_t>(1 / _keyFrameRatio.filtered() + 0.5); } else { @@ -145,13 +147,14 @@ FrameDropper::Leak(uint32_t inputFrameRate) if (_keyFrameCount > 0) { // Perform the key frame compensation - if (_keyFrameRatio.Value() > 0 && 1 / _keyFrameRatio.Value() < _keyFrameSpreadFrames) + if (_keyFrameRatio.filtered() > 0 && + 1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames) { - T -= _keyFrameSizeAvgKbits.Value() * _keyFrameRatio.Value(); + T -= _keyFrameSizeAvgKbits.filtered() * _keyFrameRatio.filtered(); } else { - T -= _keyFrameSizeAvgKbits.Value() / _keyFrameSpreadFrames; + T -= _keyFrameSizeAvgKbits.filtered() / _keyFrameSpreadFrames; } _keyFrameCount--; } @@ -232,11 +235,11 @@ FrameDropper::DropFrame() _dropCount = 0; } - if (_dropRatio.Value() >= 0.5f) // Drops per keep + if (_dropRatio.filtered() >= 0.5f) // Drops per keep { // limit is the number of frames we should drop between each kept frame // to keep our drop ratio. limit is positive in this case. - float denom = 1.0f - _dropRatio.Value(); + float denom = 1.0f - _dropRatio.filtered(); if (denom < 1e-5) { denom = (float)1e-5; @@ -252,7 +255,7 @@ FrameDropper::DropFrame() if (_dropCount < 0) { // Reset the _dropCount since it was negative and should be positive. - if (_dropRatio.Value() > 0.4f) + if (_dropRatio.filtered() > 0.4f) { _dropCount = -_dropCount; } @@ -274,12 +277,13 @@ FrameDropper::DropFrame() return false; } } - else if (_dropRatio.Value() > 0.0f && _dropRatio.Value() < 0.5f) // Keeps per drop + else if (_dropRatio.filtered() > 0.0f && + _dropRatio.filtered() < 0.5f) // Keeps per drop { // limit is the number of frames we should keep between each drop // in order to keep the drop ratio. limit is negative in this case, // and the _dropCount is also negative. - float denom = _dropRatio.Value(); + float denom = _dropRatio.filtered(); if (denom < 1e-5) { denom = (float)1e-5; @@ -289,7 +293,7 @@ FrameDropper::DropFrame() { // Reset the _dropCount since we have a positive // _dropCount, and it should be negative. - if (_dropRatio.Value() < 0.6f) + if (_dropRatio.filtered() < 0.6f) { _dropCount = -_dropCount; } @@ -350,7 +354,7 @@ FrameDropper::ActualFrameRate(uint32_t inputFrameRate) const { return static_cast<float>(inputFrameRate); } - return inputFrameRate * (1.0f - _dropRatio.Value()); + return inputFrameRate * (1.0f - _dropRatio.filtered()); } // Put a cap on the accumulator, i.e., don't let it grow beyond some level. diff --git a/modules/video_coding/utility/include/exp_filter.h b/modules/video_coding/utility/include/exp_filter.h deleted file mode 100644 index d8c37a30..00000000 --- a/modules/video_coding/utility/include/exp_filter.h +++ /dev/null @@ -1,58 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_EXP_FILTER_H_ -#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_EXP_FILTER_H_ - -namespace webrtc -{ - -/**********************/ -/* ExpFilter class */ -/**********************/ - -class VCMExpFilter -{ -public: - VCMExpFilter(float alpha, float max = -1.0) : _alpha(alpha), _filtered(-1.0), _max(max) {} - - // Resets the filter to its initial state, and resets alpha to the given value - // - // Input: - // - alpha : the new value of the filter factor base. - void Reset(float alpha); - - // Applies the filter with the given exponent on the provided sample - // - // Input: - // - exp : Exponent T in y(k) = alpha^T * y(k-1) + (1 - alpha^T) * x(k) - // - sample : x(k) in the above filter equation - float Apply(float exp, float sample); - - // Return current filtered value: y(k) - // - // Return value : The current filter output - float Value() const; - - // Change the filter factor base - // - // Input: - // - alpha : The new filter factor base. - void UpdateBase(float alpha); - -private: - float _alpha; // Filter factor base - float _filtered; // Current filter output - const float _max; -}; // end of ExpFilter class - -} // namespace webrtc - -#endif // WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_EXP_FILTER_H_ diff --git a/modules/video_coding/utility/include/frame_dropper.h b/modules/video_coding/utility/include/frame_dropper.h index 4c1c168c..8eebd784 100644 --- a/modules/video_coding/utility/include/frame_dropper.h +++ b/modules/video_coding/utility/include/frame_dropper.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_FRAME_DROPPER_H_ #define WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_FRAME_DROPPER_H_ -#include "webrtc/modules/video_coding/utility/include/exp_filter.h" +#include "webrtc/base/exp_filter.h" #include "webrtc/typedefs.h" namespace webrtc @@ -72,23 +72,23 @@ private: void UpdateRatio(); void CapAccumulator(); - VCMExpFilter _keyFrameSizeAvgKbits; - VCMExpFilter _keyFrameRatio; - float _keyFrameSpreadFrames; - int32_t _keyFrameCount; - float _accumulator; - float _accumulatorMax; - float _targetBitRate; - bool _dropNext; - VCMExpFilter _dropRatio; - int32_t _dropCount; - float _windowSize; - float _incoming_frame_rate; - bool _wasBelowMax; - bool _enabled; - bool _fastMode; - float _cap_buffer_size; - float _max_time_drops; + rtc::ExpFilter _keyFrameSizeAvgKbits; + rtc::ExpFilter _keyFrameRatio; + float _keyFrameSpreadFrames; + int32_t _keyFrameCount; + float _accumulator; + float _accumulatorMax; + float _targetBitRate; + bool _dropNext; + rtc::ExpFilter _dropRatio; + int32_t _dropCount; + float _windowSize; + float _incoming_frame_rate; + bool _wasBelowMax; + bool _enabled; + bool _fastMode; + float _cap_buffer_size; + float _max_time_drops; }; // end of VCMFrameDropper class } // namespace webrtc diff --git a/modules/video_coding/utility/video_coding_utility.gyp b/modules/video_coding/utility/video_coding_utility.gyp index 24f88800..2f0202b1 100644 --- a/modules/video_coding/utility/video_coding_utility.gyp +++ b/modules/video_coding/utility/video_coding_utility.gyp @@ -18,9 +18,7 @@ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', ], 'sources': [ - 'include/exp_filter.h', 'include/frame_dropper.h', - 'exp_filter.cc', 'frame_dropper.cc', ], }, diff --git a/modules/video_processing/BUILD.gn b/modules/video_processing/BUILD.gn index 724a9dcb..f9412f9e 100644 --- a/modules/video_processing/BUILD.gn +++ b/modules/video_processing/BUILD.gn @@ -25,8 +25,6 @@ source_set("video_processing") { "main/source/content_analysis.h", "main/source/deflickering.cc", "main/source/deflickering.h", - "main/source/denoising.cc", - "main/source/denoising.h", "main/source/frame_preprocessor.cc", "main/source/frame_preprocessor.h", "main/source/spatial_resampler.cc", diff --git a/modules/video_processing/OWNERS b/modules/video_processing/OWNERS index d5ae8473..037de93f 100644 --- a/modules/video_processing/OWNERS +++ b/modules/video_processing/OWNERS @@ -1 +1,6 @@ +stefan@webrtc.org +mikhal@webrtc.org +marpan@webrtc.org +henrik.lundin@webrtc.org + per-file BUILD.gn=kjellander@webrtc.org diff --git a/modules/video_processing/main/OWNERS b/modules/video_processing/main/OWNERS deleted file mode 100644 index 7183cf21..00000000 --- a/modules/video_processing/main/OWNERS +++ /dev/null @@ -1,4 +0,0 @@ -stefan@webrtc.org -mikhal@webrtc.org -marpan@webrtc.org -henrik.lundin@webrtc.org diff --git a/modules/video_processing/main/interface/video_processing.h b/modules/video_processing/main/interface/video_processing.h index 817d43d9..4df8812a 100644 --- a/modules/video_processing/main/interface/video_processing.h +++ b/modules/video_processing/main/interface/video_processing.h @@ -177,17 +177,6 @@ class VideoProcessingModule : public Module { virtual int32_t Deflickering(I420VideoFrame* frame, FrameStats* stats) = 0; /** - Denoises a video frame. Every frame from the stream should be passed in. - Has a fixed-point implementation. - - \param[in,out] frame - Pointer to the video frame. - - \return The number of modified pixels on success, -1 on failure. - */ - virtual int32_t Denoising(I420VideoFrame* frame) = 0; - - /** Detects if a video frame is excessively bright or dark. Returns a warning if this is the case. Multiple frames should be passed in before expecting a warning. Has a floating-point implementation. diff --git a/modules/video_processing/main/source/Android.mk b/modules/video_processing/main/source/Android.mk index 829fa960..62eb3874 100644 --- a/modules/video_processing/main/source/Android.mk +++ b/modules/video_processing/main/source/Android.mk @@ -23,7 +23,6 @@ LOCAL_SRC_FILES := \ color_enhancement.cc \ content_analysis.cc \ deflickering.cc \ - denoising.cc \ frame_preprocessor.cc \ spatial_resampler.cc \ video_decimator.cc \ diff --git a/modules/video_processing/main/source/denoising.cc b/modules/video_processing/main/source/denoising.cc deleted file mode 100644 index 4c8dcb43..00000000 --- a/modules/video_processing/main/source/denoising.cc +++ /dev/null @@ -1,156 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/video_processing/main/source/denoising.h" - -#include <string.h> - -namespace webrtc { -// Down-sampling in time (unit: number of frames) -enum { kSubsamplingTime = 0 }; -// Sub-sampling in width (unit: power of 2. -enum { kSubsamplingWidth = 0 }; -// Sub-sampling in height (unit: power of 2) -enum { kSubsamplingHeight = 0 }; -// (Q8) De-noising filter parameter -enum { kDenoiseFiltParam = 179 }; -// (Q8) 1 - filter parameter -enum { kDenoiseFiltParamRec = 77 }; -// (Q8) De-noising threshold level -enum { kDenoiseThreshold = 19200 }; - -VPMDenoising::VPMDenoising() - : id_(0), - moment1_(NULL), - moment2_(NULL) { - Reset(); -} - -VPMDenoising::~VPMDenoising() { - if (moment1_) { - delete [] moment1_; - moment1_ = NULL; -} - - if (moment2_) { - delete [] moment2_; - moment2_ = NULL; - } -} - -int32_t VPMDenoising::ChangeUniqueId(const int32_t id) { - id_ = id; - return VPM_OK; -} - -void VPMDenoising::Reset() { - frame_size_ = 0; - denoise_frame_cnt_ = 0; - - if (moment1_) { - delete [] moment1_; - moment1_ = NULL; - } - - if (moment2_) { - delete [] moment2_; - moment2_ = NULL; - } -} - -int32_t VPMDenoising::ProcessFrame(I420VideoFrame* frame) { - assert(frame); - int32_t thevar; - int k; - int jsub, ksub; - int32_t diff0; - uint32_t tmp_moment1; - uint32_t tmp_moment2; - uint32_t tmp; - int32_t num_pixels_changed = 0; - - if (frame->IsZeroSize()) { - return VPM_GENERAL_ERROR; - } - - int width = frame->width(); - int height = frame->height(); - - /* Size of luminance component */ - const uint32_t y_size = height * width; - - /* Initialization */ - if (y_size != frame_size_) { - delete [] moment1_; - moment1_ = NULL; - - delete [] moment2_; - moment2_ = NULL; - } - frame_size_ = y_size; - - if (!moment1_) { - moment1_ = new uint32_t[y_size]; - memset(moment1_, 0, sizeof(uint32_t)*y_size); - } - - if (!moment2_) { - moment2_ = new uint32_t[y_size]; - memset(moment2_, 0, sizeof(uint32_t)*y_size); - } - - /* Apply de-noising on each pixel, but update variance sub-sampled */ - uint8_t* buffer = frame->buffer(kYPlane); - for (int i = 0; i < height; i++) { // Collect over height - k = i * width; - ksub = ((i >> kSubsamplingHeight) << kSubsamplingHeight) * width; - for (int j = 0; j < width; j++) { // Collect over width - jsub = ((j >> kSubsamplingWidth) << kSubsamplingWidth); - /* Update mean value for every pixel and every frame */ - tmp_moment1 = moment1_[k + j]; - tmp_moment1 *= kDenoiseFiltParam; // Q16 - tmp_moment1 += ((kDenoiseFiltParamRec * ((uint32_t)buffer[k + j])) << 8); - tmp_moment1 >>= 8; // Q8 - moment1_[k + j] = tmp_moment1; - - tmp_moment2 = moment2_[ksub + jsub]; - if ((ksub == k) && (jsub == j) && (denoise_frame_cnt_ == 0)) { - tmp = ((uint32_t)buffer[k + j] * - (uint32_t)buffer[k + j]); - tmp_moment2 *= kDenoiseFiltParam; // Q16 - tmp_moment2 += ((kDenoiseFiltParamRec * tmp) << 8); - tmp_moment2 >>= 8; // Q8 - } - moment2_[k + j] = tmp_moment2; - /* Current event = deviation from mean value */ - diff0 = ((int32_t)buffer[k + j] << 8) - moment1_[k + j]; - /* Recent events = variance (variations over time) */ - thevar = moment2_[k + j]; - thevar -= ((moment1_[k + j] * moment1_[k + j]) >> 8); - // De-noising criteria, i.e., when should we replace a pixel by its mean. - // 1) recent events are minor. - // 2) current events are minor. - if ((thevar < kDenoiseThreshold) - && ((diff0 * diff0 >> 8) < kDenoiseThreshold)) { - // Replace with mean. - buffer[k + j] = (uint8_t)(moment1_[k + j] >> 8); - num_pixels_changed++; - } - } - } - - denoise_frame_cnt_++; - if (denoise_frame_cnt_ > kSubsamplingTime) - denoise_frame_cnt_ = 0; - - return num_pixels_changed; -} - -} // namespace diff --git a/modules/video_processing/main/source/denoising.h b/modules/video_processing/main/source/denoising.h deleted file mode 100644 index 60645fbd..00000000 --- a/modules/video_processing/main/source/denoising.h +++ /dev/null @@ -1,42 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_DENOISING_H_ -#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_DENOISING_H_ - -#include "webrtc/modules/video_processing/main/interface/video_processing.h" -#include "webrtc/typedefs.h" - -namespace webrtc { - -class VPMDenoising { - public: - VPMDenoising(); - ~VPMDenoising(); - - int32_t ChangeUniqueId(int32_t id); - - void Reset(); - - int32_t ProcessFrame(I420VideoFrame* frame); - - private: - int32_t id_; - - uint32_t* moment1_; // (Q8) First order moment (mean). - uint32_t* moment2_; // (Q8) Second order moment. - uint32_t frame_size_; // Size (# of pixels) of frame. - int denoise_frame_cnt_; // Counter for subsampling in time. -}; - -} // namespace webrtc - -#endif // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_DENOISING_H_ - diff --git a/modules/video_processing/main/source/video_processing.gypi b/modules/video_processing/main/source/video_processing.gypi index 7a0279e7..f62aa18a 100644 --- a/modules/video_processing/main/source/video_processing.gypi +++ b/modules/video_processing/main/source/video_processing.gypi @@ -31,8 +31,6 @@ 'content_analysis.h', 'deflickering.cc', 'deflickering.h', - 'denoising.cc', - 'denoising.h', 'frame_preprocessor.cc', 'frame_preprocessor.h', 'spatial_resampler.cc', diff --git a/modules/video_processing/main/source/video_processing_impl.cc b/modules/video_processing/main/source/video_processing_impl.cc index 3560030c..8bc5bf0d 100644 --- a/modules/video_processing/main/source/video_processing_impl.cc +++ b/modules/video_processing/main/source/video_processing_impl.cc @@ -51,7 +51,6 @@ int32_t VideoProcessingModuleImpl::ChangeUniqueId(const int32_t id) { id_ = id; brightness_detection_.ChangeUniqueId(id); deflickering_.ChangeUniqueId(id); - denoising_.ChangeUniqueId(id); frame_pre_processor_.ChangeUniqueId(id); return VPM_OK; } @@ -66,7 +65,6 @@ VideoProcessingModuleImpl::VideoProcessingModuleImpl(const int32_t id) mutex_(*CriticalSectionWrapper::CreateCriticalSection()) { brightness_detection_.ChangeUniqueId(id); deflickering_.ChangeUniqueId(id); - denoising_.ChangeUniqueId(id); frame_pre_processor_.ChangeUniqueId(id); } @@ -77,7 +75,6 @@ VideoProcessingModuleImpl::~VideoProcessingModuleImpl() { void VideoProcessingModuleImpl::Reset() { CriticalSectionScoped mutex(&mutex_); deflickering_.Reset(); - denoising_.Reset(); brightness_detection_.Reset(); frame_pre_processor_.Reset(); } @@ -146,11 +143,6 @@ int32_t VideoProcessingModuleImpl::Deflickering(I420VideoFrame* frame, return deflickering_.ProcessFrame(frame, stats); } -int32_t VideoProcessingModuleImpl::Denoising(I420VideoFrame* frame) { - CriticalSectionScoped mutex(&mutex_); - return denoising_.ProcessFrame(frame); -} - int32_t VideoProcessingModuleImpl::BrightnessDetection( const I420VideoFrame& frame, const FrameStats& stats) { diff --git a/modules/video_processing/main/source/video_processing_impl.h b/modules/video_processing/main/source/video_processing_impl.h index deae6ff6..6fe617d8 100644 --- a/modules/video_processing/main/source/video_processing_impl.h +++ b/modules/video_processing/main/source/video_processing_impl.h @@ -16,7 +16,6 @@ #include "webrtc/modules/video_processing/main/source/brightness_detection.h" #include "webrtc/modules/video_processing/main/source/color_enhancement.h" #include "webrtc/modules/video_processing/main/source/deflickering.h" -#include "webrtc/modules/video_processing/main/source/denoising.h" #include "webrtc/modules/video_processing/main/source/frame_preprocessor.h" namespace webrtc { @@ -36,8 +35,6 @@ class VideoProcessingModuleImpl : public VideoProcessingModule { virtual int32_t Deflickering(I420VideoFrame* frame, FrameStats* stats); - virtual int32_t Denoising(I420VideoFrame* frame); - virtual int32_t BrightnessDetection(const I420VideoFrame& frame, const FrameStats& stats); @@ -74,7 +71,6 @@ class VideoProcessingModuleImpl : public VideoProcessingModule { int32_t id_; CriticalSectionWrapper& mutex_; VPMDeflickering deflickering_; - VPMDenoising denoising_; VPMBrightnessDetection brightness_detection_; VPMFramePreprocessor frame_pre_processor_; }; diff --git a/modules/video_processing/main/test/unit_test/denoising_test.cc b/modules/video_processing/main/test/unit_test/denoising_test.cc deleted file mode 100644 index c00db6ab..00000000 --- a/modules/video_processing/main/test/unit_test/denoising_test.cc +++ /dev/null @@ -1,136 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include <stdio.h> -#include <stdlib.h> - -#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" -#include "webrtc/modules/video_processing/main/interface/video_processing.h" -#include "webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h" -#include "webrtc/system_wrappers/interface/tick_util.h" -#include "webrtc/test/testsupport/fileutils.h" -#include "webrtc/test/testsupport/gtest_disable.h" - -namespace webrtc { - -TEST_F(VideoProcessingModuleTest, DISABLED_ON_ANDROID(Denoising)) -{ - enum { NumRuns = 10 }; - uint32_t frameNum = 0; - - int64_t min_runtime = 0; - int64_t avg_runtime = 0; - - const std::string denoise_filename = - webrtc::test::OutputPath() + "denoise_testfile.yuv"; - FILE* denoiseFile = fopen(denoise_filename.c_str(), "wb"); - ASSERT_TRUE(denoiseFile != NULL) << - "Could not open output file: " << denoise_filename << "\n"; - - const std::string noise_filename = - webrtc::test::OutputPath() + "noise_testfile.yuv"; - FILE* noiseFile = fopen(noise_filename.c_str(), "wb"); - ASSERT_TRUE(noiseFile != NULL) << - "Could not open noisy file: " << noise_filename << "\n"; - - printf("\nRun time [us / frame]:\n"); - for (uint32_t run_idx = 0; run_idx < NumRuns; run_idx++) - { - TickTime t0; - TickTime t1; - TickInterval acc_ticks; - int32_t modifiedPixels = 0; - - frameNum = 0; - scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]); - while (fread(video_buffer.get(), 1, frame_length_, source_file_) == - frame_length_) - { - EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, - width_, height_, - 0, kRotateNone, &video_frame_)); - frameNum++; - uint8_t* sourceBuffer = video_frame_.buffer(kYPlane); - - // Add noise to a part in video stream - // Random noise - // TODO: investigate the effectiveness of this test. - - for (int ir = 0; ir < height_; ir++) - { - uint32_t ik = ir * width_; - for (int ic = 0; ic < width_; ic++) - { - uint8_t r = rand() % 16; - r -= 8; - if (ir < height_ / 4) - r = 0; - if (ir >= 3 * height_ / 4) - r = 0; - if (ic < width_ / 4) - r = 0; - if (ic >= 3 * width_ / 4) - r = 0; - - /*uint8_t pixelValue = 0; - if (ir >= height_ / 2) - { // Region 3 or 4 - pixelValue = 170; - } - if (ic >= width_ / 2) - { // Region 2 or 4 - pixelValue += 85; - } - pixelValue += r; - sourceBuffer[ik + ic] = pixelValue; - */ - sourceBuffer[ik + ic] += r; - } - } - - if (run_idx == 0) - { - if (PrintI420VideoFrame(video_frame_, noiseFile) < 0) { - return; - } - } - - t0 = TickTime::Now(); - ASSERT_GE(modifiedPixels = vpm_->Denoising(&video_frame_), 0); - t1 = TickTime::Now(); - acc_ticks += (t1 - t0); - - if (run_idx == 0) - { - if (PrintI420VideoFrame(video_frame_, noiseFile) < 0) { - return; - } - } - } - ASSERT_NE(0, feof(source_file_)) << "Error reading source file"; - - printf("%u\n", static_cast<int>(acc_ticks.Microseconds() / frameNum)); - if (acc_ticks.Microseconds() < min_runtime || run_idx == 0) - { - min_runtime = acc_ticks.Microseconds(); - } - avg_runtime += acc_ticks.Microseconds(); - - rewind(source_file_); - } - ASSERT_EQ(0, fclose(denoiseFile)); - ASSERT_EQ(0, fclose(noiseFile)); - printf("\nAverage run time = %d us / frame\n", - static_cast<int>(avg_runtime / frameNum / NumRuns)); - printf("Min run time = %d us / frame\n\n", - static_cast<int>(min_runtime / frameNum)); -} - -} // namespace webrtc diff --git a/modules/video_processing/main/test/unit_test/video_processing_unittest.cc b/modules/video_processing/main/test/unit_test/video_processing_unittest.cc index 973552c8..b1245038 100644 --- a/modules/video_processing/main/test/unit_test/video_processing_unittest.cc +++ b/modules/video_processing/main/test/unit_test/video_processing_unittest.cc @@ -82,8 +82,6 @@ TEST_F(VideoProcessingModuleTest, HandleNullBuffer) { EXPECT_EQ(-1, vpm_->Deflickering(&videoFrame, &stats)); - EXPECT_EQ(-1, vpm_->Denoising(&videoFrame)); - EXPECT_EQ(-3, vpm_->BrightnessDetection(videoFrame, stats)); } @@ -113,8 +111,6 @@ TEST_F(VideoProcessingModuleTest, HandleBadSize) { EXPECT_EQ(-1, vpm_->Deflickering(&video_frame_, &stats)); - EXPECT_EQ(-1, vpm_->Denoising(&video_frame_)); - EXPECT_EQ(-3, vpm_->BrightnessDetection(video_frame_, stats)); EXPECT_EQ(VPM_PARAMETER_ERROR, vpm_->SetTargetResolution(0,0,0)); @@ -145,19 +141,6 @@ TEST_F(VideoProcessingModuleTest, IdenticalResultsAfterReset) { ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_, source_file_)); - // Using ConvertToI420 to add stride to the image. - EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, - width_, height_, - 0, kRotateNone, &video_frame_)); - video_frame2.CopyFrame(video_frame_); - EXPECT_TRUE(CompareFrames(video_frame_, video_frame2)); - ASSERT_GE(vpm_->Denoising(&video_frame_), 0); - vpm_->Reset(); - ASSERT_GE(vpm_->Denoising(&video_frame2), 0); - EXPECT_TRUE(CompareFrames(video_frame_, video_frame2)); - - ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_, - source_file_)); EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_, 0, kRotateNone, &video_frame_)); |