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Код Задачи Пакеты Проекты Релизы Вики Активность

Bug 1397793 - Delete old-deprecated VoEExternalMedia. r=dminor 

MozReview-Commit-ID: C2189mFvWiY

--HG--
extra : rebase_source : 310304d83b883d82146fb4cc8422cc5c7ba080be
extra : source : b44e089bcfe7a6e2cd6dddce616c4f195d3c5ea2
This commit is contained in:
Paul Adenot 2017-10-31 17:17:20 +01:00
Родитель d6d12510a8
Коммит e4a5c78e55
6 изменённых файлов: 25 добавлений и 415 удалений
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56
media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
Просмотреть файл

@ -74,13 +74,6 @@ WebrtcAudioConduit::~WebrtcAudioConduit()
delete codec;
}
// The first one of a pair to be deleted shuts down media for both
if(mPtrVoEXmedia)
{
mPtrVoEXmedia->SetExternalRecordingStatus(false);
mPtrVoEXmedia->SetExternalPlayoutStatus(false);
}
//Deal with the transport
if(mPtrVoENetwork)
{
@ -357,8 +350,9 @@ MediaConduitErrorCode WebrtcAudioConduit::Init()
return kMediaConduitSessionNotInited;
}
// init the engine with our audio device layer
if(mPtrVoEBase->Init() == -1)
// Init the engine with a fake audio device (we're using cubeb for audio input
// and output anyways).
if(mPtrVoEBase->Init(mFakeAudioDevice.get()) == -1)
{
CSFLogError(LOGTAG, "%s VoiceEngine Base Not Initialized", __FUNCTION__);
return kMediaConduitSessionNotInited;
@ -423,21 +417,7 @@ MediaConduitErrorCode WebrtcAudioConduit::Init()
return kMediaConduitTransportRegistrationFail;
}
if(mPtrVoEXmedia->SetExternalRecordingStatus(true) == -1)
{
CSFLogError(LOGTAG, "%s SetExternalRecordingStatus Failed %d",__FUNCTION__,
mPtrVoEBase->LastError());
return kMediaConduitExternalPlayoutError;
}
if(mPtrVoEXmedia->SetExternalPlayoutStatus(true) == -1)
{
CSFLogError(LOGTAG, "%s SetExternalPlayoutStatus Failed %d ",__FUNCTION__,
mPtrVoEBase->LastError());
return kMediaConduitExternalRecordingError;
}
CSFLogDebug(LOGTAG , "%s AudioSessionConduit Initialization Done (%p)",__FUNCTION__, this);
CSFLogDebug(LOGTAG, "%s AudioSessionConduit Initialization Done (%p)",__FUNCTION__, this);
return kMediaConduitNoError;
}
@ -717,7 +697,7 @@ WebrtcAudioConduit::SendAudioFrame(const int16_t audio_data[],
}
capture_delay = mCaptureDelay;
//Insert the samples
// Insert the samples
mPtrVoEBase->audio_transport()->PushCaptureData(mChannel, audio_data,
sizeof(audio_data[0])*8, // bits
samplingFreqHz,
@ -729,9 +709,9 @@ WebrtcAudioConduit::SendAudioFrame(const int16_t audio_data[],
MediaConduitErrorCode
WebrtcAudioConduit::GetAudioFrame(int16_t speechData[],
int32_t samplingFreqHz,
int32_t capture_delay,
int& lengthSamples)
int32_t samplingFreqHz,
int32_t capture_delay,
int& lengthSamples)
{
CSFLogDebug(LOGTAG, "%s ", __FUNCTION__);
@ -772,11 +752,9 @@ WebrtcAudioConduit::GetAudioFrame(int16_t speechData[],
lengthSamples = 0; //output paramter
if(mPtrVoEXmedia->ExternalPlayoutGetData( speechData,
samplingFreqHz,
capture_delay,
lengthSamples) == -1)
{
if (mPtrVoEXmedia->GetAudioFrame(mChannel,
samplingFreqHz,
&mAudioFrame) != 0) {
int error = mPtrVoEBase->LastError();
CSFLogError(LOGTAG, "%s Getting audio data Failed %d", __FUNCTION__, error);
if(error == VE_RUNTIME_PLAY_ERROR)
@ -786,6 +764,11 @@ WebrtcAudioConduit::GetAudioFrame(int16_t speechData[],
return kMediaConduitUnknownError;
}
// XXX Annoying, have to copy to our buffers -- refactor?
lengthSamples = mAudioFrame.samples_per_channel_ * mAudioFrame.num_channels_;
PodCopy(speechData, mAudioFrame.data_,
lengthSamples);
// Not #ifdef DEBUG or on a log module so we can use it for about:webrtc/etc
mSamples += lengthSamples;
if (mSamples >= mLastSyncLog + samplingFreqHz) {
@ -964,6 +947,13 @@ WebrtcAudioConduit::StartReceiving()
return kMediaConduitUnknownError;
}
// we can't call GetAudioFrame() if we don't enable "external" mixing
if(mPtrVoEXmedia->SetExternalMixing(mChannel, true) == -1)
{
CSFLogError(LOGTAG, "%s SetExternalMixing Failed", __FUNCTION__);
return kMediaConduitPlayoutError;
}
if(mPtrVoEBase->StartPlayout(mChannel) == -1)
{
CSFLogError(LOGTAG, "%s Starting playout Failed", __FUNCTION__);

5
media/webrtc/signaling/src/peerconnection/TransceiverImpl.cpp
Просмотреть файл

@ -1020,8 +1020,7 @@ static void StartTrack(MediaStream* aSource,
segment_(aSegment) {}
void Run() override {
TrackRate track_rate = segment_->GetType() == MediaSegment::AUDIO ?
WEBRTC_DEFAULT_SAMPLE_RATE : mStream->GraphRate();
TrackRate track_rate = mStream->GraphRate();
StreamTime current_end = mStream->GetTracksEnd();
MOZ_MTLOG(ML_DEBUG, "current_end = " << current_end);
TrackTicks current_ticks =
@ -1043,7 +1042,7 @@ static void StartTrack(MediaStream* aSource,
MOZ_MTLOG(ML_DEBUG, "Calling AddAudioTrack");
mStream->AsSourceStream()->AddAudioTrack(
kAudioTrack,
WEBRTC_DEFAULT_SAMPLE_RATE,
track_rate,
0,
static_cast<AudioSegment*>(segment_.forget()));
} else {

53
media/webrtc/trunk/webrtc/voice_engine/include/voe_external_media.h
Просмотреть файл

@ -8,27 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
// In some cases it is desirable to use an audio source or sink which may
// not be available to the VoiceEngine, such as a DV camera. This sub-API
// contains functions that allow for the use of such external recording
// sources and playout sinks. It also describes how recorded data, or data
// to be played out, can be modified outside the VoiceEngine.
//
// Usage example, omitting error checking:
//
// using namespace webrtc;
// VoiceEngine* voe = VoiceEngine::Create();
// VoEBase* base = VoEBase::GetInterface(voe);
// VoEMediaProcess media = VoEMediaProcess::GetInterface(voe);
// base->Init();
// ...
// media->SetExternalRecordingStatus(true);
// ...
// base->Terminate();
// base->Release();
// media->Release();
// VoiceEngine::Delete(voe);
//
#ifndef WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
#define WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
@ -85,38 +64,6 @@ class WEBRTC_DLLEXPORT VoEExternalMedia {
// media for the specified |channel| and |type|.
virtual int DeRegisterExternalMediaProcessing(int channel,
ProcessingTypes type) = 0;
// Toogles state of external recording.
virtual int SetExternalRecordingStatus(bool enable) = 0;
// Toogles state of external playout.
virtual int SetExternalPlayoutStatus(bool enable) = 0;
// This function accepts externally recorded audio. During transmission,
// this method should be called at as regular an interval as possible
// with frames of corresponding size.
virtual int ExternalRecordingInsertData(
const int16_t speechData10ms[], int lengthSamples,
int samplingFreqHz, int current_delay_ms) = 0;
// This function inserts audio written to the OS audio drivers for use
// as the far-end signal for AEC processing. The length of the block
// must be 160, 320, 441 or 480 samples (for 16000, 32000, 44100 or
// 48000 kHz sampling rates respectively).
virtual int ExternalPlayoutData(
int16_t speechData10ms[], int samplingFreqHz, int num_channels,
int& lengthSamples) = 0;
// This function gets audio for an external playout sink.
// During transmission, this function should be called every ~10 ms
// to obtain a new 10 ms frame of audio. The length of the block will
// be 160, 320, 441 or 480 samples (for 16000, 32000, 44100 or
// 48000 kHz sampling rates respectively).
virtual int ExternalPlayoutGetData(
int16_t speechData10ms[], int samplingFreqHz,
int current_delay_ms, int& lengthSamples) = 0;
// Pulls an audio frame from the specified |channel| for external mixing.
// If the |desired_sample_rate_hz| is 0, the signal will be returned with
// its native frequency, otherwise it will be resampled. Valid frequencies

37
media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
Просмотреть файл

@ -29,43 +29,6 @@ class ExternalMediaTest : public AfterStreamingFixture {
}
};
TEST_F(ExternalMediaTest, ManualCanRecordAndPlaybackUsingExternalPlayout) {
SwitchToManualMicrophone();
EXPECT_EQ(0, voe_base_->StopSend(channel_));
EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
EXPECT_EQ(0, voe_xmedia_->SetExternalPlayoutStatus(true));
EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
EXPECT_EQ(0, voe_base_->StartSend(channel_));
TEST_LOG("Recording data for 2 seconds starting now: please speak.\n");
int16_t recording[32000];
for (int i = 0; i < 200; i++) {
int sample_length = 0;
EXPECT_EQ(0, voe_xmedia_->ExternalPlayoutGetData(
&(recording[i * 160]), 16000, 100, sample_length));
EXPECT_EQ(160, sample_length);
Sleep(10);
}
EXPECT_EQ(0, voe_base_->StopSend(channel_));
EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
EXPECT_EQ(0, voe_xmedia_->SetExternalPlayoutStatus(false));
EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
EXPECT_EQ(0, voe_xmedia_->SetExternalRecordingStatus(true));
EXPECT_EQ(0, voe_base_->StartSend(channel_));
TEST_LOG("Playing back recording, you should hear what you said earlier.\n");
for (int i = 0; i < 200; i++) {
EXPECT_EQ(0, voe_xmedia_->ExternalRecordingInsertData(
&(recording[i * 160]), 160, 16000, 20));
Sleep(10);
}
EXPECT_EQ(0, voe_base_->StopSend(channel_));
EXPECT_EQ(0, voe_xmedia_->SetExternalRecordingStatus(false));
}
TEST_F(ExternalMediaTest,
ManualRegisterExternalMediaProcessingOnAllChannelsAffectsPlayout) {
TEST_LOG("Enabling external media processing: audio should be affected.\n");

264
media/webrtc/trunk/webrtc/voice_engine/voe_external_media_impl.cc
Просмотреть файл

@ -31,9 +31,6 @@ VoEExternalMedia* VoEExternalMedia::GetInterface(VoiceEngine* voiceEngine) {
VoEExternalMediaImpl::VoEExternalMediaImpl(voe::SharedData* shared)
:
#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
playout_delay_ms_(0),
#endif
shared_(shared) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(shared_->instance_id(), -1),
"VoEExternalMediaImpl() - ctor");
@ -116,267 +113,6 @@ int VoEExternalMediaImpl::DeRegisterExternalMediaProcessing(
return -1;
}
int VoEExternalMediaImpl::SetExternalRecordingStatus(bool enable)
{
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(shared_->instance_id(), -1),
"SetExternalRecordingStatus(enable=%d)", enable);
#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
if (shared_->audio_device() && shared_->audio_device()->Recording())
{
shared_->SetLastError(VE_ALREADY_SENDING, kTraceError,
"SetExternalRecordingStatus() cannot set state while sending");
return -1;
}
shared_->set_ext_recording(enable);
return 0;
#else
shared_->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
"SetExternalRecordingStatus() external recording is not supported");
return -1;
#endif
}
int VoEExternalMediaImpl::ExternalRecordingInsertData(
const int16_t speechData10ms[],
int lengthSamples,
int samplingFreqHz,
int current_delay_ms)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(shared_->instance_id(), -1),
"ExternalRecordingInsertData(speechData10ms=0x%x,"
" lengthSamples=%u, samplingFreqHz=%d, current_delay_ms=%d)",
&speechData10ms[0], lengthSamples, samplingFreqHz,
current_delay_ms);
#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
if (!shared_->statistics().Initialized())
{
shared_->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
if (!shared_->ext_recording())
{
shared_->SetLastError(VE_INVALID_OPERATION, kTraceError,
"ExternalRecordingInsertData() external recording is not enabled");
return -1;
}
if (shared_->NumOfSendingChannels() == 0)
{
shared_->SetLastError(VE_ALREADY_SENDING, kTraceError,
"SetExternalRecordingStatus() no channel is sending");
return -1;
}
if ((16000 != samplingFreqHz) && (32000 != samplingFreqHz) &&
(48000 != samplingFreqHz) && (44100 != samplingFreqHz))
{
shared_->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
"SetExternalRecordingStatus() invalid sample rate");
return -1;
}
if ((0 == lengthSamples) ||
((lengthSamples % (samplingFreqHz / 100)) != 0))
{
shared_->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
"SetExternalRecordingStatus() invalid buffer size");
return -1;
}
if (current_delay_ms < 0)
{
shared_->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
"SetExternalRecordingStatus() invalid delay)");
return -1;
}
uint16_t blockSize = samplingFreqHz / 100;
// We know the number of samples for 10ms of audio, so we can derive the
// number of channels here:
uint32_t channels = lengthSamples * 100 / samplingFreqHz;
uint32_t nBlocks = lengthSamples / blockSize / channels;
int16_t totalDelayMS = 0;
uint16_t playoutDelayMS = 0;
for (uint32_t i = 0; i < nBlocks; i++)
{
if (!shared_->ext_playout())
{
// Use real playout delay if external playout is not enabled.
if (shared_->audio_device()->PlayoutDelay(&playoutDelayMS) != 0) {
shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
"PlayoutDelay() unable to get the playout delay");
}
totalDelayMS = current_delay_ms + playoutDelayMS;
}
else
{
// Use stored delay value given the last call
// to ExternalPlayoutGetData.
totalDelayMS = current_delay_ms + playout_delay_ms_;
// Compensate for block sizes larger than 10ms
totalDelayMS -= (int16_t)(i*10);
if (totalDelayMS < 0)
totalDelayMS = 0;
}
shared_->transmit_mixer()->PrepareDemux(
(const int8_t*)(&speechData10ms[i*blockSize]),
blockSize,
channels,
samplingFreqHz,
totalDelayMS,
0,
0,
false); // Typing detection not supported
shared_->transmit_mixer()->DemuxAndMix();
shared_->transmit_mixer()->EncodeAndSend();
}
return 0;
#else
shared_->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
"ExternalRecordingInsertData() external recording is not supported");
return -1;
#endif
}
int VoEExternalMediaImpl::SetExternalPlayoutStatus(bool enable)
{
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(shared_->instance_id(), -1),
"SetExternalPlayoutStatus(enable=%d)", enable);
#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
if (shared_->audio_device() && shared_->audio_device()->Playing())
{
shared_->SetLastError(VE_ALREADY_SENDING, kTraceError,
"SetExternalPlayoutStatus() cannot set state while playing");
return -1;
}
shared_->set_ext_playout(enable);
return 0;
#else
shared_->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
"SetExternalPlayoutStatus() external playout is not supported");
return -1;
#endif
}
// This inserts a copy of the raw audio sent to the output drivers to use
// as the "far end" signal for the AEC. Currently only 10ms chunks are
// supported unfortunately. Since we have to rechunk to 10ms to call this,
// thre isn't much gained by allowing N*10ms here; external code can loop
// if needed.
int VoEExternalMediaImpl::ExternalPlayoutData(
int16_t speechData10ms[],
int samplingFreqHz,
int num_channels,
int& lengthSamples)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(shared_->instance_id(), -1),
"ExternalPlayoutData(speechData10ms=0x%x,"
" lengthSamples=%u, samplingFreqHz=%d)",
&speechData10ms[0], lengthSamples, samplingFreqHz);
#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
if (!shared_->statistics().Initialized())
{
shared_->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
// FIX(jesup) - check if this is enabled?
if (shared_->NumOfSendingChannels() == 0)
{
shared_->SetLastError(VE_ALREADY_SENDING, kTraceError,
"SetExternalRecordingStatus() no channel is sending");
return -1;
}
if ((16000 != samplingFreqHz) && (32000 != samplingFreqHz) &&
(48000 != samplingFreqHz) && (44100 != samplingFreqHz))
{
shared_->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
"SetExternalRecordingStatus() invalid sample rate");
return -1;
}
// Far-end data is inserted without going through neteq/etc.
// Only supports 10ms chunks; AnalyzeReverseStream() enforces that
// lower down.
AudioFrame audioFrame;
audioFrame.UpdateFrame(-1, 0xFFFFFFFF,
speechData10ms,
lengthSamples,
samplingFreqHz,
AudioFrame::kNormalSpeech,
AudioFrame::kVadUnknown,
num_channels);
shared_->output_mixer()->APMAnalyzeReverseStream(audioFrame);
#endif
return 0;
}
int VoEExternalMediaImpl::ExternalPlayoutGetData(
int16_t speechData10ms[],
int samplingFreqHz,
int current_delay_ms,
int& lengthSamples)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(shared_->instance_id(), -1),
"ExternalPlayoutGetData(speechData10ms=0x%x, samplingFreqHz=%d"
", current_delay_ms=%d)", &speechData10ms[0], samplingFreqHz,
current_delay_ms);
#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
if (!shared_->statistics().Initialized())
{
shared_->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
if (!shared_->ext_playout())
{
shared_->SetLastError(VE_INVALID_OPERATION, kTraceError,
"ExternalPlayoutGetData() external playout is not enabled");
return -1;
}
if ((16000 != samplingFreqHz) && (32000 != samplingFreqHz) &&
(48000 != samplingFreqHz) && (44100 != samplingFreqHz))
{
shared_->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
"ExternalPlayoutGetData() invalid sample rate");
return -1;
}
if (current_delay_ms < 0)
{
shared_->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
"ExternalPlayoutGetData() invalid delay)");
return -1;
}
AudioFrame audioFrame;
uint32_t channels = shared_->output_mixer()->GetOutputChannelCount();
// If we have not received any data yet, consider it's mono since it's the
// most common case.
if (channels == 0) {
channels = 1;
}
// Retrieve mixed output at the specified rate
shared_->output_mixer()->MixActiveChannels();
shared_->output_mixer()->DoOperationsOnCombinedSignal(true);
shared_->output_mixer()->GetMixedAudio(samplingFreqHz, channels, &audioFrame);
// Deliver audio (PCM) samples to the external sink
memcpy(speechData10ms,
audioFrame.data_,
sizeof(int16_t)*audioFrame.samples_per_channel_*channels);
lengthSamples = audioFrame.samples_per_channel_ * channels;
// Store current playout delay (to be used by ExternalRecordingInsertData).
playout_delay_ms_ = current_delay_ms;
return 0;
#else
shared_->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
"ExternalPlayoutGetData() external playout is not supported");
return -1;
#endif
}
int VoEExternalMediaImpl::GetAudioFrame(int channel, int desired_sample_rate_hz,
AudioFrame* frame) {
if (!shared_->statistics().Initialized()) {

25
media/webrtc/trunk/webrtc/voice_engine/voe_external_media_impl.h
Просмотреть файл

@ -26,28 +26,6 @@ class VoEExternalMediaImpl : public VoEExternalMedia {
int DeRegisterExternalMediaProcessing(int channel,
ProcessingTypes type) override;
virtual int SetExternalRecordingStatus(bool enable) override;
virtual int SetExternalPlayoutStatus(bool enable) override;
virtual int ExternalRecordingInsertData(
const int16_t speechData10ms[],
int lengthSamples,
int samplingFreqHz,
int current_delay_ms) override;
// Insertion of far-end data as actually played out to the OS audio driver
virtual int ExternalPlayoutData(
int16_t speechData10ms[],
int samplingFreqHz,
int num_channels,
int& lengthSamples) override;
virtual int ExternalPlayoutGetData(int16_t speechData10ms[],
int samplingFreqHz,
int current_delay_ms,
int& lengthSamples) override;
int GetAudioFrame(int channel,
int desired_sample_rate_hz,
AudioFrame* frame) override;
@ -59,9 +37,6 @@ class VoEExternalMediaImpl : public VoEExternalMedia {
~VoEExternalMediaImpl() override;
private:
#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
int playout_delay_ms_;
#endif
voe::SharedData* shared_;
};