aboutsummaryrefslogtreecommitdiff
path: root/webrtc/modules/audio_coding/neteq/test
diff options
context:
space:
mode:
Diffstat (limited to 'webrtc/modules/audio_coding/neteq/test')
-rw-r--r--webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h2
-rw-r--r--webrtc/modules/audio_coding/neteq/test/PayloadTypes.h2
-rw-r--r--webrtc/modules/audio_coding/neteq/test/RTPencode.cc63
-rw-r--r--webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc10
-rw-r--r--webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc2
-rw-r--r--webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc2
-rw-r--r--webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc10
7 files changed, 68 insertions, 23 deletions
diff --git a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
index 3fbce8be5c..56ed72fcee 100644
--- a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
+++ b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
@@ -14,7 +14,7 @@
#include <map>
#include <stdio.h>
#include "webrtc/typedefs.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
enum stereoModes {
stereoModeMono,
diff --git a/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h b/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h
index c46a3daece..aba525b162 100644
--- a/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h
+++ b/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h
@@ -39,7 +39,7 @@
#define NETEQ_CODEC_G722_1_16_PT 108
#define NETEQ_CODEC_G722_1_24_PT 109
#define NETEQ_CODEC_G722_1_32_PT 110
-#define NETEQ_CODEC_SC3_PT 111
+#define NETEQ_CODEC_OPUS_PT 111
#define NETEQ_CODEC_AMR_PT 112
#define NETEQ_CODEC_GSMEFR_PT 113
//#define NETEQ_CODEC_ILBCRCU_PT 114
diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
index cbb7436152..45586ee111 100644
--- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
@@ -25,7 +25,9 @@
#include <algorithm>
+#include "webrtc/base/checks.h"
#include "webrtc/typedefs.h"
+
// needed for NetEqDecoder
#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
@@ -36,6 +38,10 @@
#include "PayloadTypes.h"
+namespace {
+const size_t kRtpDataSize = 8000;
+}
+
/*********************/
/* Misc. definitions */
/*********************/
@@ -126,10 +132,10 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride);
#include "webrtc_vad.h"
#if ((defined CODEC_PCM16B) || (defined NETEQ_ARBITRARY_CODEC))
-#include "pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
#endif
#ifdef CODEC_G711
-#include "g711_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
#endif
#ifdef CODEC_G729
#include "G729Interface.h"
@@ -146,19 +152,19 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride);
#include "AMRWBCreation.h"
#endif
#ifdef CODEC_ILBC
-#include "ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
#endif
#if (defined CODEC_ISAC || defined CODEC_ISAC_SWB)
-#include "isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
#endif
#ifdef NETEQ_ISACFIX_CODEC
-#include "isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
#ifdef CODEC_ISAC
#error Cannot have both ISAC and ISACfix defined. Please de-select one.
#endif
#endif
#ifdef CODEC_G722
-#include "g722_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
#endif
#ifdef CODEC_G722_1_24
#include "G722_1Interface.h"
@@ -188,11 +194,14 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride);
#endif
#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
-#include "webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
#endif
#if ((defined CODEC_SPEEX_8) || (defined CODEC_SPEEX_16))
#include "SpeexInterface.h"
#endif
+#ifdef CODEC_OPUS
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#endif
/***********************************/
/* Global codec instance variables */
@@ -264,6 +273,9 @@ SPEEX_encinst_t* SPEEX8enc_inst[2];
#ifdef CODEC_SPEEX_16
SPEEX_encinst_t* SPEEX16enc_inst[2];
#endif
+#ifdef CODEC_OPUS
+OpusEncInst* opus_inst[2];
+#endif
int main(int argc, char* argv[]) {
size_t packet_size;
@@ -275,7 +287,7 @@ int main(int argc, char* argv[]) {
int useRed = 0;
size_t len, enc_len;
int16_t org_data[4000];
- unsigned char rtp_data[8000];
+ unsigned char rtp_data[kRtpDataSize];
int16_t seqNo = 0xFFF;
uint32_t ssrc = 1235412312;
uint32_t timestamp = 0xAC1245;
@@ -286,12 +298,12 @@ int main(int argc, char* argv[]) {
uint32_t red_TS[2] = {0};
uint16_t red_len[2] = {0};
size_t RTPheaderLen = 12;
- uint8_t red_data[8000];
+ uint8_t red_data[kRtpDataSize];
#ifdef INSERT_OLD_PACKETS
uint16_t old_length, old_plen;
size_t old_enc_len;
int first_old_packet = 1;
- unsigned char old_rtp_data[8000];
+ unsigned char old_rtp_data[kRtpDataSize];
size_t packet_age = 0;
#endif
#ifdef INSERT_DTMF_PACKETS
@@ -429,6 +441,10 @@ int main(int argc, char* argv[]) {
printf(" : red_isac Redundancy RTP packet with 2*iSAC "
"frames\n");
#endif
+#endif // CODEC_RED
+#ifdef CODEC_OPUS
+ printf(" : opus Opus codec with FEC (48kHz, 32kbps, FEC"
+ " on and tuned for 5%% packet losses)\n");
#endif
printf("\n");
@@ -880,6 +896,10 @@ void NetEQTest_GetCodec_and_PT(char* name,
*PT = NETEQ_CODEC_ISAC_PT; /* this will be the PT for the sub-headers */
*fs = 16000;
*useRed = 1;
+ } else if (!strcmp(name, "opus")) {
+ *codec = webrtc::NetEqDecoder::kDecoderOpus;
+ *PT = NETEQ_CODEC_OPUS_PT; /* this will be the PT for the sub-headers */
+ *fs = 48000;
} else {
printf("Error: Not a supported codec (%s)\n", name);
exit(0);
@@ -1411,12 +1431,23 @@ int NetEQTest_init_coders(webrtc::NetEqDecoder coder,
}
break;
#endif
+#ifdef CODEC_OPUS
+ case webrtc::NetEqDecoder::kDecoderOpus:
+ ok = WebRtcOpus_EncoderCreate(&opus_inst[k], 1, 0);
+ if (ok != 0) {
+ printf("Error: Couldn't allocate memory for Opus encoding "
+ "instance\n");
+ exit(0);
+ }
+ WebRtcOpus_EnableFec(opus_inst[k]);
+ WebRtcOpus_SetPacketLossRate(opus_inst[k], 5);
+ break;
+#endif
default:
printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
exit(0);
break;
}
-
if (ok != 0) {
return (ok);
}
@@ -1543,6 +1574,11 @@ int NetEQTest_free_coders(webrtc::NetEqDecoder coder, size_t numChannels) {
WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]);
break;
#endif
+#ifdef CODEC_OPUS
+ case webrtc::NetEqDecoder::kDecoderOpus:
+ WebRtcOpus_EncoderFree(opus_inst[k]);
+ break;
+#endif
default:
printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
exit(0);
@@ -1687,6 +1723,11 @@ size_t NetEQTest_encode(webrtc::NetEqDecoder coder,
cdlen = static_cast<size_t>(res);
}
#endif
+#ifdef CODEC_OPUS
+ cdlen = WebRtcOpus_Encode(opus_inst[k], indata, frameLen, kRtpDataSize - 12,
+ encoded);
+ RTC_CHECK_GT(cdlen, 0u);
+#endif
indata += frameLen;
encoded += cdlen;
totalLen += cdlen;
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
index 2042e0d2b8..0c09e92b4d 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
@@ -11,7 +11,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "webrtc/test/testsupport/fileutils.h"
@@ -50,7 +50,7 @@ class NetEqIlbcQualityTest : public NetEqQualityTest {
NetEqDecoder::kDecoderILBC) {}
void SetUp() override {
- ASSERT_EQ(1, channels_) << "iLBC supports only mono audio.";
+ ASSERT_EQ(1u, channels_) << "iLBC supports only mono audio.";
AudioEncoderIlbc::Config config;
config.frame_size_ms = FLAGS_frame_size_ms;
encoder_.reset(new AudioEncoderIlbc(config));
@@ -66,8 +66,10 @@ class NetEqIlbcQualityTest : public NetEqQualityTest {
uint32_t dummy_timestamp = 0;
AudioEncoder::EncodedInfo info;
do {
- info = encoder_->Encode(dummy_timestamp, &in_data[encoded_samples],
- kFrameSizeSamples, max_bytes, payload);
+ info = encoder_->Encode(dummy_timestamp,
+ rtc::ArrayView<const int16_t>(
+ in_data + encoded_samples, kFrameSizeSamples),
+ max_bytes, payload);
encoded_samples += kFrameSizeSamples;
} while (info.encoded_bytes == 0);
return rtc::checked_cast<int>(info.encoded_bytes);
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
index 66b0903f66..4ccebb3e66 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
@@ -59,7 +59,7 @@ NetEqIsacQualityTest::NetEqIsacQualityTest()
bit_rate_kbps_(FLAGS_bit_rate_kbps) {}
void NetEqIsacQualityTest::SetUp() {
- ASSERT_EQ(1, channels_) << "iSAC supports only mono audio.";
+ ASSERT_EQ(1u, channels_) << "iSAC supports only mono audio.";
// Create encoder memory.
WebRtcIsacfix_Create(&isac_encoder_);
ASSERT_TRUE(isac_encoder_ != NULL);
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
index 5e8b2297d4..5ab55ba9e8 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
#include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
index 422a9fa6eb..ac478ab5ac 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
@@ -11,7 +11,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "webrtc/test/testsupport/fileutils.h"
@@ -50,7 +50,7 @@ class NetEqPcmuQualityTest : public NetEqQualityTest {
NetEqDecoder::kDecoderPCMu) {}
void SetUp() override {
- ASSERT_EQ(1, channels_) << "PCMu supports only mono audio.";
+ ASSERT_EQ(1u, channels_) << "PCMu supports only mono audio.";
AudioEncoderPcmU::Config config;
config.frame_size_ms = FLAGS_frame_size_ms;
encoder_.reset(new AudioEncoderPcmU(config));
@@ -66,8 +66,10 @@ class NetEqPcmuQualityTest : public NetEqQualityTest {
uint32_t dummy_timestamp = 0;
AudioEncoder::EncodedInfo info;
do {
- info = encoder_->Encode(dummy_timestamp, &in_data[encoded_samples],
- kFrameSizeSamples, max_bytes, payload);
+ info = encoder_->Encode(dummy_timestamp,
+ rtc::ArrayView<const int16_t>(
+ in_data + encoded_samples, kFrameSizeSamples),
+ max_bytes, payload);
encoded_samples += kFrameSizeSamples;
} while (info.encoded_bytes == 0);
return rtc::checked_cast<int>(info.encoded_bytes);
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-29 12:43:00 +0000