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Bug 953265: Update webrtc/getUserMedia default audio capture rate to 32KHz r=padenot

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This commit is contained in:
Randell Jesup 2015-09-24 09:23:37 -04:00
parent 7ae59e77ce
commit 46bfe823f7
5 changed files with 23 additions and 9 deletions
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6
dom/media/webrtc/MediaEngine.h
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@ -54,6 +54,12 @@ public:
static const int DEFAULT_169_VIDEO_HEIGHT = 720;
static const int DEFAULT_AUDIO_TIMER_MS = 10;
#ifndef MOZ_B2G
static const int DEFAULT_SAMPLE_RATE = 32000;
#else
static const int DEFAULT_SAMPLE_RATE = 16000;
#endif
/* Populate an array of video sources in the nsTArray. Also include devices
* that are currently unavailable. */
virtual void EnumerateVideoDevices(dom::MediaSourceEnum,

2
dom/media/webrtc/MediaEngineDefault.cpp
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@ -26,7 +26,7 @@
#include "YuvStamper.h"
#endif
#define AUDIO_RATE 16000
#define AUDIO_RATE mozilla::MediaEngine::DEFAULT_SAMPLE_RATE
#define AUDIO_FRAME_LENGTH ((AUDIO_RATE * MediaEngine::DEFAULT_AUDIO_TIMER_MS) / 1000)
namespace mozilla {

2
dom/media/webrtc/MediaEngineWebRTC.h
View File

@ -137,6 +137,7 @@ public:
, mChannel(-1)
, mInitDone(false)
, mStarted(false)
, mSampleFrequency(MediaEngine::DEFAULT_SAMPLE_RATE)
, mEchoOn(false), mAgcOn(false), mNoiseOn(false)
, mEchoCancel(webrtc::kEcDefault)
, mAGC(webrtc::kAgcDefault)
@ -225,6 +226,7 @@ private:
nsString mDeviceName;
nsCString mDeviceUUID;
uint32_t mSampleFrequency;
bool mEchoOn, mAgcOn, mNoiseOn;
webrtc::EcModes mEchoCancel;
webrtc::AgcModes mAGC;

17
dom/media/webrtc/MediaEngineWebRTCAudio.cpp
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@ -19,9 +19,8 @@
#define ENCODING "L16"
#define DEFAULT_PORT 5555
#define SAMPLE_RATE 256000
#define SAMPLE_FREQUENCY 16000
#define SAMPLE_LENGTH ((SAMPLE_FREQUENCY*10)/1000)
#define SAMPLE_RATE(freq) ((freq)*2*8) // bps, 16-bit samples
#define SAMPLE_LENGTH(freq) (((freq)*10)/1000)
// These are restrictions from the webrtc.org code
#define MAX_CHANNELS 2
@ -345,7 +344,7 @@ MediaEngineWebRTCMicrophoneSource::Start(SourceMediaStream *aStream,
}
AudioSegment* segment = new AudioSegment();
aStream->AddAudioTrack(aID, SAMPLE_FREQUENCY, 0, segment, SourceMediaStream::ADDTRACK_QUEUED);
aStream->AddAudioTrack(aID, mSampleFrequency, 0, segment, SourceMediaStream::ADDTRACK_QUEUED);
// XXX Make this based on the pref.
aStream->RegisterForAudioMixing();
@ -470,6 +469,9 @@ MediaEngineWebRTCMicrophoneSource::Init()
return;
}
mSampleFrequency = MediaEngine::DEFAULT_SAMPLE_RATE;
LOG(("%s: sampling rate %u", __FUNCTION__, mSampleFrequency));
// Check for availability.
ScopedCustomReleasePtr<webrtc::VoEHardware> ptrVoEHw(webrtc::VoEHardware::GetInterface(mVoiceEngine));
if (!ptrVoEHw || ptrVoEHw->SetRecordingDevice(mCapIndex)) {
@ -495,9 +497,10 @@ MediaEngineWebRTCMicrophoneSource::Init()
webrtc::CodecInst codec;
strcpy(codec.plname, ENCODING);
codec.channels = CHANNELS;
codec.rate = SAMPLE_RATE;
codec.plfreq = SAMPLE_FREQUENCY;
codec.pacsize = SAMPLE_LENGTH;
MOZ_ASSERT(mSampleFrequency == 16000 || mSampleFrequency == 32000);
codec.rate = SAMPLE_RATE(mSampleFrequency);
codec.plfreq = mSampleFrequency;
codec.pacsize = SAMPLE_LENGTH(mSampleFrequency);
codec.pltype = 0; // Default payload type
if (!ptrVoECodec->SetSendCodec(mChannel, codec)) {

5
media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
View File

@ -48,6 +48,9 @@
#include "logging.h"
// Should come from MediaEngineWebRTC.h, but that's a pain to include here
#define DEFAULT_SAMPLE_RATE 32000
using namespace mozilla;
using namespace mozilla::gfx;
using namespace mozilla::layers;
@ -1332,7 +1335,7 @@ void GenericReceiveListener::AddSelf(MediaSegment* segment) {
MediaPipelineReceiveAudio::PipelineListener::PipelineListener(
SourceMediaStream * source, TrackID track_id,
const RefPtr<MediaSessionConduit>& conduit, bool queue_track)
: GenericReceiveListener(source, track_id, 16000, queue_track), // XXX rate assumption
: GenericReceiveListener(source, track_id, DEFAULT_SAMPLE_RATE, queue_track), // XXX rate assumption
conduit_(conduit)
{
MOZ_ASSERT(track_rate_%100 == 0);