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-rw-r--r--webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc6
-rw-r--r--webrtc/modules/audio_processing/test/unit_test.cc26
-rw-r--r--webrtc/modules/interface/module_common_types.h84
-rw-r--r--webrtc/modules/utility/source/audio_frame_operations_unittest.cc9
-rw-r--r--webrtc/modules/utility/source/coder.cc3
-rw-r--r--webrtc/voice_engine/channel.cc22
-rw-r--r--webrtc/voice_engine/output_mixer.cc12
-rw-r--r--webrtc/voice_engine/output_mixer_internal.cc4
-rw-r--r--webrtc/voice_engine/output_mixer_unittest.cc4
-rw-r--r--webrtc/voice_engine/transmit_mixer.cc3
-rw-r--r--webrtc/voice_engine/voe_file_impl.cc129
11 files changed, 120 insertions, 182 deletions
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
index fb5203746f..3fedc487ac 100644
--- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
+++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
@@ -173,7 +173,7 @@ bool AudioConferenceMixerImpl::Init()
if (!SetNumLimiterChannels(1))
return false;
- if(_limiter->gain_control()->set_mode(GainControl::kFixedDigital) !=
+ if(_limiter->gain_control()->set_mode(GainControl::kFixedDigital) !=
_limiter->kNoError)
return false;
@@ -1125,7 +1125,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::MixFromList(
{
// No mixing required here; skip the saturation protection.
AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem());
- mixedAudio = *audioFrame;
+ mixedAudio.CopyFrom(*audioFrame);
SetParticipantStatistics(&_scratchMixedParticipants[position],
*audioFrame);
return 0;
@@ -1173,7 +1173,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::MixAnonomouslyFromList(
{
// No mixing required here; skip the saturation protection.
AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem());
- mixedAudio = *audioFrame;
+ mixedAudio.CopyFrom(*audioFrame);
return 0;
}
diff --git a/webrtc/modules/audio_processing/test/unit_test.cc b/webrtc/modules/audio_processing/test/unit_test.cc
index 51340e959a..1baa48d67c 100644
--- a/webrtc/modules/audio_processing/test/unit_test.cc
+++ b/webrtc/modules/audio_processing/test/unit_test.cc
@@ -451,7 +451,8 @@ void ApmTest::ChangeTriggersInit(F f, AudioProcessing* ap, int initial_value,
EnableAllComponents();
Init(16000, 2, 2, 2, false);
SetFrameTo(frame_, 1000);
- AudioFrame frame_copy = *frame_;
+ AudioFrame frame_copy;
+ frame_copy.CopyFrom(*frame_);
ProcessWithDefaultStreamParameters(frame_);
// Verify the processing has actually changed the frame.
EXPECT_FALSE(FrameDataAreEqual(*frame_, frame_copy));
@@ -464,13 +465,14 @@ void ApmTest::ChangeTriggersInit(F f, AudioProcessing* ap, int initial_value,
apm_->Initialize();
SetFrameTo(frame_, 1000);
- AudioFrame initial_frame = *frame_;
+ AudioFrame initial_frame;
+ initial_frame.CopyFrom(*frame_);
ProcessWithDefaultStreamParameters(frame_);
ProcessWithDefaultStreamParameters(frame_);
// Verify the processing has actually changed the frame.
EXPECT_FALSE(FrameDataAreEqual(*frame_, initial_frame));
- frame_copy = initial_frame;
+ frame_copy.CopyFrom(initial_frame);
apm_->Initialize();
ProcessWithDefaultStreamParameters(&frame_copy);
// Verify an init here would result in different output.
@@ -478,7 +480,7 @@ void ApmTest::ChangeTriggersInit(F f, AudioProcessing* ap, int initial_value,
ProcessWithDefaultStreamParameters(&frame_copy);
EXPECT_FALSE(FrameDataAreEqual(*frame_, frame_copy));
- frame_copy = initial_frame;
+ frame_copy.CopyFrom(initial_frame);
apm_->Initialize();
ProcessWithDefaultStreamParameters(&frame_copy);
// Test that the same value does not trigger an init.
@@ -1078,7 +1080,8 @@ TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabled) {
for (size_t i = 0; i < kSampleRatesSize; i++) {
Init(kSampleRates[i], 2, 2, 2, false);
SetFrameTo(frame_, 1000, 2000);
- AudioFrame frame_copy = *frame_;
+ AudioFrame frame_copy;
+ frame_copy.CopyFrom(*frame_);
for (int j = 0; j < 1000; j++) {
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
@@ -1119,14 +1122,15 @@ TEST_F(ApmTest, SplittingFilter) {
// Verify the filter is not active through undistorted audio when:
// 1. No components are enabled...
SetFrameTo(frame_, 1000);
- AudioFrame frame_copy = *frame_;
+ AudioFrame frame_copy;
+ frame_copy.CopyFrom(*frame_);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
// 2. Only the level estimator is enabled...
SetFrameTo(frame_, 1000);
- frame_copy = *frame_;
+ frame_copy.CopyFrom(*frame_);
EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
@@ -1135,7 +1139,7 @@ TEST_F(ApmTest, SplittingFilter) {
// 3. Only VAD is enabled...
SetFrameTo(frame_, 1000);
- frame_copy = *frame_;
+ frame_copy.CopyFrom(*frame_);
EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
@@ -1144,7 +1148,7 @@ TEST_F(ApmTest, SplittingFilter) {
// 4. Both VAD and the level estimator are enabled...
SetFrameTo(frame_, 1000);
- frame_copy = *frame_;
+ frame_copy.CopyFrom(*frame_);
EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
@@ -1164,7 +1168,7 @@ TEST_F(ApmTest, SplittingFilter) {
// behavior of the AEC. Think of something more robust.
EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
SetFrameTo(frame_, 1000);
- frame_copy = *frame_;
+ frame_copy.CopyFrom(*frame_);
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
@@ -1182,7 +1186,7 @@ TEST_F(ApmTest, SplittingFilter) {
frame_->num_channels_ = 2;
frame_->sample_rate_hz_ = 32000;
SetFrameTo(frame_, 1000);
- frame_copy = *frame_;
+ frame_copy.CopyFrom(*frame_);
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
diff --git a/webrtc/modules/interface/module_common_types.h b/webrtc/modules/interface/module_common_types.h
index 4b0dc5a8d4..a33e7efb07 100644
--- a/webrtc/modules/interface/module_common_types.h
+++ b/webrtc/modules/interface/module_common_types.h
@@ -11,8 +11,10 @@
#ifndef MODULE_COMMON_TYPES_H
#define MODULE_COMMON_TYPES_H
+#include <cassert>
#include <cstring> // memcpy
-#include <assert.h>
+
+#include <algorithm>
#include "webrtc/common_types.h"
#include "webrtc/system_wrappers/interface/constructor_magic.h"
@@ -724,7 +726,8 @@ VideoFrame::Free()
class AudioFrame
{
public:
- enum { kMaxDataSizeSamples = 3840 }; // stereo, 32 kHz, 60ms (2*32*60)
+ // Stereo, 32 kHz, 60 ms (2 * 32 * 60)
+ static const int kMaxDataSizeSamples = 3840;
enum VADActivity
{
@@ -744,7 +747,7 @@ public:
AudioFrame();
virtual ~AudioFrame();
- int UpdateFrame(
+ void UpdateFrame(
int id,
uint32_t timestamp,
const int16_t* data,
@@ -757,9 +760,10 @@ public:
AudioFrame& Append(const AudioFrame& rhs);
+ void CopyFrom(const AudioFrame& src);
+
void Mute();
- AudioFrame& operator=(const AudioFrame& rhs);
AudioFrame& operator>>=(const int rhs);
AudioFrame& operator+=(const AudioFrame& rhs);
AudioFrame& operator-=(const AudioFrame& rhs);
@@ -773,6 +777,9 @@ public:
SpeechType speech_type_;
VADActivity vad_activity_;
uint32_t energy_;
+
+private:
+ DISALLOW_COPY_AND_ASSIGN(AudioFrame);
};
inline
@@ -796,7 +803,7 @@ AudioFrame::~AudioFrame()
}
inline
-int
+void
AudioFrame::UpdateFrame(
int id,
uint32_t timestamp,
@@ -810,67 +817,50 @@ AudioFrame::UpdateFrame(
{
id_ = id;
timestamp_ = timestamp;
+ samples_per_channel_ = samples_per_channel;
sample_rate_hz_ = sample_rate_hz;
speech_type_ = speech_type;
vad_activity_ = vad_activity;
num_channels_ = num_channels;
energy_ = energy;
- if((samples_per_channel > kMaxDataSizeSamples) ||
- (num_channels > 2) || (num_channels < 1))
- {
- samples_per_channel_ = 0;
- return -1;
- }
- samples_per_channel_ = samples_per_channel;
+ const int length = samples_per_channel * num_channels;
+ assert(length <= kMaxDataSizeSamples && length >= 0);
if(data != NULL)
{
- memcpy(data_, data, sizeof(int16_t) *
- samples_per_channel * num_channels_);
+ memcpy(data_, data, sizeof(int16_t) * length);
}
else
{
- memset(data_,0,sizeof(int16_t) *
- samples_per_channel * num_channels_);
+ memset(data_, 0, sizeof(int16_t) * length);
}
- return 0;
}
-inline
-void
-AudioFrame::Mute()
+inline void AudioFrame::CopyFrom(const AudioFrame& src)
{
- memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t));
+ if(this == &src)
+ {
+ return;
+ }
+ id_ = src.id_;
+ timestamp_ = src.timestamp_;
+ samples_per_channel_ = src.samples_per_channel_;
+ sample_rate_hz_ = src.sample_rate_hz_;
+ speech_type_ = src.speech_type_;
+ vad_activity_ = src.vad_activity_;
+ num_channels_ = src.num_channels_;
+ energy_ = src.energy_;
+
+ const int length = samples_per_channel_ * num_channels_;
+ assert(length <= kMaxDataSizeSamples && length >= 0);
+ memcpy(data_, src.data_, sizeof(int16_t) * length);
}
inline
-AudioFrame&
-AudioFrame::operator=(const AudioFrame& rhs)
+void
+AudioFrame::Mute()
{
- // Sanity Check
- if((rhs.samples_per_channel_ > kMaxDataSizeSamples) ||
- (rhs.num_channels_ > 2) ||
- (rhs.num_channels_ < 1))
- {
- return *this;
- }
- if(this == &rhs)
- {
- return *this;
- }
- id_ = rhs.id_;
- timestamp_ = rhs.timestamp_;
- sample_rate_hz_ = rhs.sample_rate_hz_;
- speech_type_ = rhs.speech_type_;
- vad_activity_ = rhs.vad_activity_;
- num_channels_ = rhs.num_channels_;
- energy_ = rhs.energy_;
-
- samples_per_channel_ = rhs.samples_per_channel_;
- memcpy(data_, rhs.data_,
- sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
-
- return *this;
+ memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t));
}
inline
diff --git a/webrtc/modules/utility/source/audio_frame_operations_unittest.cc b/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
index c11dce3cab..efdebf6dd2 100644
--- a/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
+++ b/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
@@ -62,7 +62,8 @@ TEST_F(AudioFrameOperationsTest, MonoToStereoFailsWithBadParameters) {
TEST_F(AudioFrameOperationsTest, MonoToStereoSucceeds) {
frame_.num_channels_ = 1;
SetFrameData(&frame_, 1);
- AudioFrame temp_frame = frame_;
+ AudioFrame temp_frame;
+ temp_frame.CopyFrom(frame_);
EXPECT_EQ(0, AudioFrameOperations::MonoToStereo(&frame_));
AudioFrame stereo_frame;
@@ -86,7 +87,8 @@ TEST_F(AudioFrameOperationsTest, StereoToMonoFailsWithBadParameters) {
TEST_F(AudioFrameOperationsTest, StereoToMonoSucceeds) {
SetFrameData(&frame_, 4, 2);
- AudioFrame temp_frame = frame_;
+ AudioFrame temp_frame;
+ temp_frame.CopyFrom(frame_);
EXPECT_EQ(0, AudioFrameOperations::StereoToMono(&frame_));
AudioFrame mono_frame;
@@ -131,7 +133,8 @@ TEST_F(AudioFrameOperationsTest, SwapStereoChannelsFailsOnMono) {
// Set data to "stereo", despite it being a mono frame.
SetFrameData(&frame_, 0, 1);
- AudioFrame orig_frame = frame_;
+ AudioFrame orig_frame;
+ orig_frame.CopyFrom(frame_);
AudioFrameOperations::SwapStereoChannels(&frame_);
// Verify that no swap occurred.
VerifyFramesAreEqual(orig_frame, frame_);
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
index 31d528ddb7..a610daa6bd 100644
--- a/webrtc/modules/utility/source/coder.cc
+++ b/webrtc/modules/utility/source/coder.cc
@@ -92,7 +92,8 @@ WebRtc_Word32 AudioCoder::Encode(const AudioFrame& audio,
{
// Fake a timestamp in case audio doesn't contain a correct timestamp.
// Make a local copy of the audio frame since audio is const
- AudioFrame audioFrame = audio;
+ AudioFrame audioFrame;
+ audioFrame.CopyFrom(audio);
audioFrame.timestamp_ = _encodeTimestamp;
_encodeTimestamp += audioFrame.samples_per_channel_;
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 38897fc0da..357eb0ffe2 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -660,7 +660,7 @@ Channel::OnInitializeDecoder(
receiveCodec.channels = channels;
receiveCodec.rate = rate;
strncpy(receiveCodec.plname, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
-
+
_audioCodingModule.Codec(payloadName, dummyCodec, frequency, channels);
receiveCodec.pacsize = dummyCodec.pacsize;
@@ -3789,7 +3789,7 @@ int Channel::StartPlayingFileAsMicrophone(InStream* stream,
_inputFilePlayerPtr = NULL;
return -1;
}
-
+
_inputFilePlayerPtr->RegisterModuleFileCallback(this);
_inputFilePlaying = true;
@@ -4009,7 +4009,7 @@ int Channel::StartRecordingPlayout(OutStream* stream,
_outputFileRecorderPtr = NULL;
return -1;
}
-
+
_outputFileRecorderPtr->RegisterModuleFileCallback(this);
_outputFileRecording = true;
@@ -5794,7 +5794,7 @@ Channel::Demultiplex(const AudioFrame& audioFrame)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::Demultiplex()");
- _audioFrame = audioFrame;
+ _audioFrame.CopyFrom(audioFrame);
_audioFrame.id_ = _channelId;
return 0;
}
@@ -6343,7 +6343,7 @@ Channel::InsertInbandDtmfTone()
// account.
_inbandDtmfGenerator.ResetTone();
}
-
+
WebRtc_Word16 toneBuffer[320];
WebRtc_UWord16 toneSamples(0);
// Get 10ms tone segment and set time since last tone to zero
@@ -6356,19 +6356,19 @@ Channel::InsertInbandDtmfTone()
}
// Replace mixed audio with DTMF tone.
- for (int sample = 0;
+ for (int sample = 0;
sample < _audioFrame.samples_per_channel_;
sample++)
{
- for (int channel = 0;
- channel < _audioFrame.num_channels_;
+ for (int channel = 0;
+ channel < _audioFrame.num_channels_;
channel++)
{
- _audioFrame.data_[sample * _audioFrame.num_channels_ + channel] =
- toneBuffer[sample];
+ const int index = sample * _audioFrame.num_channels_ + channel;
+ _audioFrame.data_[index] = toneBuffer[sample];
}
}
-
+
assert(_audioFrame.samples_per_channel_ == toneSamples);
} else
{
diff --git a/webrtc/voice_engine/output_mixer.cc b/webrtc/voice_engine/output_mixer.cc
index daf0d4a306..36a9da7345 100644
--- a/webrtc/voice_engine/output_mixer.cc
+++ b/webrtc/voice_engine/output_mixer.cc
@@ -32,7 +32,7 @@ OutputMixer::NewMixedAudio(const WebRtc_Word32 id,
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::NewMixedAudio(id=%d, size=%u)", id, size);
- _audioFrame = generalAudioFrame;
+ _audioFrame.CopyFrom(generalAudioFrame);
_audioFrame.id_ = id;
}
@@ -135,7 +135,7 @@ OutputMixer::OutputMixer(const WebRtc_UWord32 instanceId) :
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::OutputMixer() - ctor");
-
+
if ((_mixerModule.RegisterMixedStreamCallback(*this) == -1) ||
(_mixerModule.RegisterMixerStatusCallback(*this, 100) == -1))
{
@@ -143,7 +143,7 @@ OutputMixer::OutputMixer(const WebRtc_UWord32 instanceId) :
"OutputMixer::OutputMixer() failed to register mixer"
"callbacks");
}
-
+
_dtmfGenerator.Init();
}
@@ -156,7 +156,7 @@ OutputMixer::Destroy(OutputMixer*& mixer)
mixer = NULL;
}
}
-
+
OutputMixer::~OutputMixer()
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
@@ -191,7 +191,7 @@ OutputMixer::SetEngineInformation(voe::Statistics& engineStatistics)
return 0;
}
-WebRtc_Word32
+WebRtc_Word32
OutputMixer::SetAudioProcessingModule(
AudioProcessing* audioProcessingModule)
{
@@ -368,7 +368,7 @@ int OutputMixer::StartRecordingPlayout(const char* fileName,
}
CriticalSectionScoped cs(&_fileCritSect);
-
+
// Destroy the old instance
if (_outputFileRecorderPtr)
{
diff --git a/webrtc/voice_engine/output_mixer_internal.cc b/webrtc/voice_engine/output_mixer_internal.cc
index b78d8cd936..dfa7d95b8f 100644
--- a/webrtc/voice_engine/output_mixer_internal.cc
+++ b/webrtc/voice_engine/output_mixer_internal.cc
@@ -39,7 +39,7 @@ int RemixAndResample(const AudioFrame& src_frame,
if (resampler->ResetIfNeeded(src_frame.sample_rate_hz_,
dst_frame->sample_rate_hz_,
resampler_type) == -1) {
- *dst_frame = src_frame;
+ dst_frame->CopyFrom(src_frame);
WEBRTC_TRACE(kTraceError, kTraceVoice, -1,
"%s ResetIfNeeded failed", __FUNCTION__);
return -1;
@@ -53,7 +53,7 @@ int RemixAndResample(const AudioFrame& src_frame,
out_length) == 0) {
dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
} else {
- *dst_frame = src_frame;
+ dst_frame->CopyFrom(src_frame);
WEBRTC_TRACE(kTraceError, kTraceVoice, -1,
"%s resampling failed", __FUNCTION__);
return -1;
diff --git a/webrtc/voice_engine/output_mixer_unittest.cc b/webrtc/voice_engine/output_mixer_unittest.cc
index fe678a02a8..dbcb25103a 100644
--- a/webrtc/voice_engine/output_mixer_unittest.cc
+++ b/webrtc/voice_engine/output_mixer_unittest.cc
@@ -25,8 +25,8 @@ class OutputMixerTest : public ::testing::Test {
src_frame_.sample_rate_hz_ = 16000;
src_frame_.samples_per_channel_ = src_frame_.sample_rate_hz_ / 100;
src_frame_.num_channels_ = 1;
- dst_frame_ = src_frame_;
- golden_frame_ = src_frame_;
+ dst_frame_.CopyFrom(src_frame_);
+ golden_frame_.CopyFrom(src_frame_);
}
void RunResampleTest(int src_channels, int src_sample_rate_hz,
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index 4dc18498bb..a699401207 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -444,7 +444,8 @@ TransmitMixer::DemuxAndMix()
} else if (channelPtr->Sending())
{
// load temporary audioframe with current (mixed) microphone signal
- AudioFrame tmpAudioFrame = _audioFrame;
+ AudioFrame tmpAudioFrame;
+ tmpAudioFrame.CopyFrom(_audioFrame);
channelPtr->Demultiplex(tmpAudioFrame);
channelPtr->PrepareEncodeAndSend(_mixingFrequency);
diff --git a/webrtc/voice_engine/voe_file_impl.cc b/webrtc/voice_engine/voe_file_impl.cc
index 8f0061faae..869cd8640b 100644
--- a/webrtc/voice_engine/voe_file_impl.cc
+++ b/webrtc/voice_engine/voe_file_impl.cc
@@ -704,17 +704,10 @@ int VoEFileImpl::ConvertPCMToWAV(const char* fileNameInUTF8,
break;
}
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength,
- frequency, AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "ConvertPCMToWAV failed during conversion (audio frame)");
- break;
- }
+ audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength,
+ frequency, AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
res=recObj.RecordAudioToFile(audioFrame);
if(res)
@@ -794,18 +787,10 @@ int VoEFileImpl::ConvertPCMToWAV(InStream* streamIn, OutStream* streamOut)
break;
}
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength, frequency,
- AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "ConvertPCMToWAV failed during conversion "
- "(create audio frame)");
- break;
- }
+ audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength, frequency,
+ AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
res=recObj.RecordAudioToFile(audioFrame);
if(res)
@@ -882,17 +867,10 @@ int VoEFileImpl::ConvertWAVToPCM(const char* fileNameInUTF8,
break;
}
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength,
- frequency, AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "ConvertWAVToPCM failed during conversion (audio frame)");
- break;
- }
+ audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength,
+ frequency, AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
res=recObj.RecordAudioToFile(audioFrame);
if(res)
@@ -974,17 +952,10 @@ int VoEFileImpl::ConvertWAVToPCM(InStream* streamIn, OutStream* streamOut)
break;
}
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength, frequency,
- AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "ConvertWAVToPCM failed during conversion (audio frame)");
- break;
- }
+ audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength, frequency,
+ AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
res=recObj.RecordAudioToFile(audioFrame);
if(res)
@@ -1059,18 +1030,10 @@ int VoEFileImpl::ConvertPCMToCompressed(const char* fileNameInUTF8,
// This is an OK way to end
break;
}
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength,
- frequency, AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "ConvertPCMToCompressed failed during conversion "
- "(audio frame)");
- break;
- }
+ audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength,
+ frequency, AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
res=recObj.RecordAudioToFile(audioFrame);
if(res)
@@ -1151,18 +1114,10 @@ int VoEFileImpl::ConvertPCMToCompressed(InStream* streamIn,
// This is an OK way to end
break;
}
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength,
- frequency, AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "ConvertPCMToCompressed failed during conversion "
- "(audio frame)");
- break;
- }
+ audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength,
+ frequency, AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
res=recObj.RecordAudioToFile(audioFrame);
if(res)
@@ -1241,19 +1196,11 @@ int VoEFileImpl::ConvertCompressedToPCM(const char* fileNameInUTF8,
// This is an OK way to end
break;
}
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength,
- frequency,
- AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "ConvertCompressedToPCM failed during conversion "
- "(create audio frame)");
- break;
- }
+ audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength,
+ frequency,
+ AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
res=recObj.RecordAudioToFile(audioFrame);
if(res)
@@ -1338,19 +1285,11 @@ int VoEFileImpl::ConvertCompressedToPCM(InStream* streamIn,
// This is an OK way to end
break;
}
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength,
- frequency,
- AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "ConvertCompressedToPCM failed during conversion "
- "(audio frame)");
- break;
- }
+ audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength,
+ frequency,
+ AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
res=recObj.RecordAudioToFile(audioFrame);
if(res)
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-11 19:09:00 +0000