generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Properly error check calls to AudioProcessing.

Browse Source 
Checks must be made with "!= 0", not "== -1". Additionally:
* Clean up the function calling into AudioProcessing.
* Remove the unused _noiseWarning.
* Make the other warnings bool.

BUG=chromium:178040

Review URL: https://webrtc-codereview.appspot.com/1147004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3590 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org 2013-03-01 18:47:28 +00:00
parent 9ee5a4ccd8
commit 6be1e934ad
7 changed files with 67 additions and 157 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

3
webrtc/modules/audio_processing/echo_cancellation_impl.cc
View File

@ -212,10 +212,9 @@ int EchoCancellationImpl::device_sample_rate_hz() const {
return device_sample_rate_hz_;
}
int EchoCancellationImpl::set_stream_drift_samples(int drift) {
void EchoCancellationImpl::set_stream_drift_samples(int drift) {
was_stream_drift_set_ = true;
stream_drift_samples_ = drift;
return apm_->kNoError;
}
int EchoCancellationImpl::stream_drift_samples() const {

2
webrtc/modules/audio_processing/echo_cancellation_impl.h
View File

@ -41,7 +41,7 @@ class EchoCancellationImpl : public EchoCancellation,
virtual int enable_drift_compensation(bool enable);
virtual bool is_drift_compensation_enabled() const;
virtual int set_device_sample_rate_hz(int rate);
virtual int set_stream_drift_samples(int drift);
virtual void set_stream_drift_samples(int drift);
virtual int set_suppression_level(SuppressionLevel level);
virtual SuppressionLevel suppression_level() const;
virtual int enable_metrics(bool enable);

4
webrtc/modules/audio_processing/include/audio_processing.h
View File

@ -285,8 +285,8 @@ class EchoCancellation {
// Sets the difference between the number of samples rendered and captured by
// the audio devices since the last call to |ProcessStream()|. Must be called
// if and only if drift compensation is enabled, prior to |ProcessStream()|.
virtual int set_stream_drift_samples(int drift) = 0;
// if drift compensation is enabled, prior to |ProcessStream()|.
virtual void set_stream_drift_samples(int drift) = 0;
virtual int stream_drift_samples() const = 0;
enum SuppressionLevel {

6
webrtc/modules/audio_processing/test/process_test.cc
View File

@ -689,8 +689,7 @@ void void_main(int argc, char* argv[]) {
apm->gain_control()->set_stream_analog_level(msg.level()));
ASSERT_EQ(apm->kNoError,
apm->set_stream_delay_ms(msg.delay() + extra_delay_ms));
ASSERT_EQ(apm->kNoError,
apm->echo_cancellation()->set_stream_drift_samples(msg.drift()));
apm->echo_cancellation()->set_stream_drift_samples(msg.drift());
int err = apm->ProcessStream(&near_frame);
if (err == apm->kBadStreamParameterWarning) {
@ -893,8 +892,7 @@ void void_main(int argc, char* argv[]) {
apm->gain_control()->set_stream_analog_level(capture_level));
ASSERT_EQ(apm->kNoError,
apm->set_stream_delay_ms(delay_ms + extra_delay_ms));
ASSERT_EQ(apm->kNoError,
apm->echo_cancellation()->set_stream_drift_samples(drift_samples));
apm->echo_cancellation()->set_stream_drift_samples(drift_samples);
int err = apm->ProcessStream(&near_frame);
if (err == apm->kBadStreamParameterWarning) {

30
webrtc/modules/audio_processing/test/unit_test.cc
View File

@ -444,8 +444,7 @@ bool ApmTest::ReadFrame(FILE* file, AudioFrame* frame) {
void ApmTest::ProcessWithDefaultStreamParameters(AudioFrame* frame) {
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame));
@ -603,8 +602,7 @@ TEST_F(ApmTest, StreamParameters) {
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_drift_compensation(true));
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kStreamParameterNotSetError,
apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
@ -625,8 +623,7 @@ TEST_F(ApmTest, StreamParameters) {
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_drift_compensation(true));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
@ -638,8 +635,7 @@ TEST_F(ApmTest, StreamParameters) {
// Resets after successful ProcessStream().
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
@ -660,8 +656,7 @@ TEST_F(ApmTest, StreamParameters) {
// -- All there --
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
@ -1275,8 +1270,7 @@ TEST_F(ApmTest, IdenticalInputChannelsResultInIdenticalOutputChannels) {
frame_->vad_activity_ = AudioFrame::kVadUnknown;
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(analog_level));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
@ -1341,12 +1335,10 @@ TEST_F(ApmTest, SplittingFilter) {
SetFrameTo(frame_, 1000);
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));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
@ -1359,8 +1351,7 @@ TEST_F(ApmTest, SplittingFilter) {
SetFrameTo(frame_, 1000);
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));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_FALSE(FrameDataAreEqual(*frame_, frame_copy));
}
@ -1452,8 +1443,7 @@ TEST_F(ApmTest, Process) {
frame_->vad_activity_ = AudioFrame::kVadUnknown;
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(analog_level));

168
webrtc/voice_engine/transmit_mixer.cc
View File

@ -20,6 +20,7 @@
#include "utility.h"
#include "voe_base_impl.h"
#include "voe_external_media.h"
#include "webrtc/system_wrappers/interface/logging.h"
#define WEBRTC_ABS(a) (((a) < 0) ? -(a) : (a))
@ -28,10 +29,11 @@ namespace webrtc {
namespace voe {
// Used for downmixing before resampling.
// TODO(andrew): audio_device should advertise the maximum sample rate it can
// provide.
// TODO(ajm): audio_device should advertise the maximum sample rate it can
// provide.
static const int kMaxMonoDeviceDataSizeSamples = 960; // 10 ms, 96 kHz, mono.
// TODO(ajm): The thread safety of this is dubious...
void
TransmitMixer::OnPeriodicProcess()
{
@ -39,7 +41,7 @@ TransmitMixer::OnPeriodicProcess()
"TransmitMixer::OnPeriodicProcess()");
#if defined(WEBRTC_VOICE_ENGINE_TYPING_DETECTION)
if (_typingNoiseWarning > 0)
if (_typingNoiseWarning)
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_voiceEngineObserverPtr)
@ -50,11 +52,11 @@ TransmitMixer::OnPeriodicProcess()
_voiceEngineObserverPtr->CallbackOnError(-1,
VE_TYPING_NOISE_WARNING);
}
_typingNoiseWarning = 0;
_typingNoiseWarning = false;
}
#endif
if (_saturationWarning > 0)
if (_saturationWarning)
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_voiceEngineObserverPtr)
@ -63,21 +65,8 @@ TransmitMixer::OnPeriodicProcess()
"TransmitMixer::OnPeriodicProcess() =>"
" CallbackOnError(VE_SATURATION_WARNING)");
_voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING);
}
_saturationWarning = 0;
}
if (_noiseWarning > 0)
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_voiceEngineObserverPtr)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
"TransmitMixer::OnPeriodicProcess() =>"
"CallbackOnError(VE_NOISE_WARNING)");
_voiceEngineObserverPtr->CallbackOnError(-1, VE_NOISE_WARNING);
}
_noiseWarning = 0;
_saturationWarning = false;
}
}
@ -169,7 +158,7 @@ TransmitMixer::Destroy(TransmitMixer*& mixer)
TransmitMixer::TransmitMixer(const WebRtc_UWord32 instanceId) :
_engineStatisticsPtr(NULL),
_channelManagerPtr(NULL),
_audioProcessingModulePtr(NULL),
audioproc_(NULL),
_voiceEngineObserverPtr(NULL),
_processThreadPtr(NULL),
_filePlayerPtr(NULL),
@ -190,15 +179,14 @@ TransmitMixer::TransmitMixer(const WebRtc_UWord32 instanceId) :
_timeActive(0),
_timeSinceLastTyping(0),
_penaltyCounter(0),
_typingNoiseWarning(0),
_typingNoiseWarning(false),
_timeWindow(10), // 10ms slots accepted to count as a hit
_costPerTyping(100), // Penalty added for a typing + activity coincide
_reportingThreshold(300), // Threshold for _penaltyCounter
_penaltyDecay(1), // how much we reduce _penaltyCounter every 10 ms.
_typeEventDelay(2), // how "old" event we check for
#endif
_saturationWarning(0),
_noiseWarning(0),
_saturationWarning(false),
_instanceId(instanceId),
_mixFileWithMicrophone(false),
_captureLevel(0),
@ -303,7 +291,7 @@ TransmitMixer::SetAudioProcessingModule(AudioProcessing* audioProcessingModule)
"TransmitMixer::SetAudioProcessingModule("
"audioProcessingModule=0x%x)",
audioProcessingModule);
_audioProcessingModulePtr = audioProcessingModule;
audioproc_ = audioProcessingModule;
return 0;
}
@ -370,8 +358,8 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
}
}
// --- Near-end Voice Quality Enhancement (APM) processing
APMProcessStream(totalDelayMS, clockDrift, currentMicLevel);
// --- Near-end audio processing.
ProcessAudio(totalDelayMS, clockDrift, currentMicLevel);
if (swap_stereo_channels_ && stereo_codec_)
// Only bother swapping if we're using a stereo codec.
@ -1293,94 +1281,43 @@ WebRtc_Word32 TransmitMixer::MixOrReplaceAudioWithFile(
return 0;
}
WebRtc_Word32 TransmitMixer::APMProcessStream(
const WebRtc_UWord16 totalDelayMS,
const WebRtc_Word32 clockDrift,
const WebRtc_UWord16 currentMicLevel)
{
WebRtc_UWord16 captureLevel(currentMicLevel);
void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift,
int current_mic_level) {
if (audioproc_->set_num_channels(_audioFrame.num_channels_,
_audioFrame.num_channels_) != 0) {
LOG_FERR2(LS_ERROR, set_num_channels, _audioFrame.num_channels_,
_audioFrame.num_channels_);
}
// Check if the number of incoming channels has changed. This has taken
// both the capture device and send codecs into account.
if (_audioFrame.num_channels_ !=
_audioProcessingModulePtr->num_input_channels())
{
if (_audioProcessingModulePtr->set_num_channels(
_audioFrame.num_channels_,
_audioFrame.num_channels_))
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
"AudioProcessing::set_num_channels(%d, %d) => error",
_audioFrame.num_channels_,
_audioProcessingModulePtr->num_output_channels());
}
}
if (audioproc_->set_sample_rate_hz(_audioFrame.sample_rate_hz_) != 0) {
LOG_FERR1(LS_ERROR, set_sample_rate_hz, _audioFrame.sample_rate_hz_);
}
// If the frequency has changed we need to change APM settings
// Sending side is "master"
if (_audioProcessingModulePtr->sample_rate_hz() !=
_audioFrame.sample_rate_hz_)
{
if (_audioProcessingModulePtr->set_sample_rate_hz(
_audioFrame.sample_rate_hz_))
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
"AudioProcessing::set_sample_rate_hz(%u) => error",
_audioFrame.sample_rate_hz_);
}
}
if (audioproc_->set_stream_delay_ms(delay_ms) != 0) {
// Report as a warning; we can occasionally run into very large delays.
LOG_FERR1(LS_WARNING, set_stream_delay_ms, delay_ms);
}
if (_audioProcessingModulePtr->set_stream_delay_ms(totalDelayMS) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
"AudioProcessing::set_stream_delay_ms(%u) => error",
totalDelayMS);
}
if (_audioProcessingModulePtr->gain_control()->set_stream_analog_level(
captureLevel) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
"AudioProcessing::set_stream_analog_level(%u) => error",
captureLevel);
}
if (_audioProcessingModulePtr->echo_cancellation()->
is_drift_compensation_enabled())
{
if (_audioProcessingModulePtr->echo_cancellation()->
set_stream_drift_samples(clockDrift) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
"AudioProcessing::set_stream_drift_samples(%u) => error",
clockDrift);
}
}
if (_audioProcessingModulePtr->ProcessStream(&_audioFrame) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
"AudioProcessing::ProcessStream() => error");
}
captureLevel =
_audioProcessingModulePtr->gain_control()->stream_analog_level();
GainControl* agc = audioproc_->gain_control();
if (agc->set_stream_analog_level(current_mic_level) != 0) {
LOG_FERR1(LS_ERROR, set_stream_analog_level, current_mic_level);
}
// Store new capture level (only updated when analog AGC is enabled)
_captureLevel = captureLevel;
EchoCancellation* aec = audioproc_->echo_cancellation();
if (aec->is_drift_compensation_enabled()) {
aec->set_stream_drift_samples(clock_drift);
}
// Log notifications
if (_audioProcessingModulePtr->gain_control()->stream_is_saturated())
{
if (_saturationWarning == 1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
"TransmitMixer::APMProcessStream() pending "
"saturation warning exists");
}
_saturationWarning = 1; // triggers callback from moduleprocess thread
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
"TransmitMixer::APMProcessStream() VE_SATURATION_WARNING "
"message has been posted for callback");
}
int err = audioproc_->ProcessStream(&_audioFrame);
if (err != 0) {
LOG(LS_ERROR) << "ProcessStream() error: " << err;
}
return 0;
// Store new capture level. Only updated when analog AGC is enabled.
_captureLevel = agc->stream_analog_level();
// Triggers a callback in OnPeriodicProcess().
_saturationWarning |= agc->stream_is_saturated();
}
#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
@ -1422,19 +1359,8 @@ int TransmitMixer::TypingDetection()
_penaltyCounter += _costPerTyping;
if (_penaltyCounter > _reportingThreshold)
{
if (_typingNoiseWarning == 1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, -1),
"TransmitMixer::TypingDetection() pending "
"noise-saturation warning exists");
}
// triggers callback from the module process thread
_typingNoiseWarning = 1;
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
"TransmitMixer::TypingDetection() "
"VE_TYPING_NOISE_WARNING message has been posted for"
"callback");
// Triggers a callback in OnPeriodicProcess().
_typingNoiseWarning = true;
}
}

11
webrtc/voice_engine/transmit_mixer.h
View File

@ -175,9 +175,7 @@ private:
WebRtc_Word32 MixOrReplaceAudioWithFile(
const int mixingFrequency);
WebRtc_Word32 APMProcessStream(const WebRtc_UWord16 totalDelayMS,
const WebRtc_Word32 clockDrift,
const WebRtc_UWord16 currentMicLevel);
void ProcessAudio(int delay_ms, int clock_drift, int current_mic_level);
#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
int TypingDetection();
@ -186,7 +184,7 @@ private:
// uses
Statistics* _engineStatisticsPtr;
ChannelManager* _channelManagerPtr;
AudioProcessing* _audioProcessingModulePtr;
AudioProcessing* audioproc_;
VoiceEngineObserver* _voiceEngineObserverPtr;
ProcessThread* _processThreadPtr;
@ -212,7 +210,7 @@ private:
WebRtc_Word32 _timeActive;
WebRtc_Word32 _timeSinceLastTyping;
WebRtc_Word32 _penaltyCounter;
WebRtc_UWord32 _typingNoiseWarning;
bool _typingNoiseWarning;
// Tunable treshold values
int _timeWindow; // nr of10ms slots accepted to count as a hit.
@ -222,8 +220,7 @@ private:
int _typeEventDelay; // How old typing events we allow
#endif
WebRtc_UWord32 _saturationWarning;
WebRtc_UWord32 _noiseWarning;
bool _saturationWarning;
int _instanceId;
bool _mixFileWithMicrophone;