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Bug 1198588 - Remove unused MSG-specific code from AudioStream. r=padenot

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This commit is contained in:
Matthew Gregan 2015-08-27 14:49:17 +12:00
parent 7e7dda409c
commit 93c6746c0c
3 changed files with 33 additions and 540 deletions
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440
dom/media/AudioStream.cpp
View File

@ -15,12 +15,8 @@
#include "mozilla/Snprintf.h"
#include <algorithm>
#include "mozilla/Telemetry.h"
#include "Latency.h"
#include "CubebUtils.h"
#include "nsPrintfCString.h"
#ifdef XP_MACOSX
#include <sys/sysctl.h>
#endif
namespace mozilla {
@ -130,18 +126,11 @@ AudioStream::AudioStream()
, mWritten(0)
, mAudioClock(this)
, mTimeStretcher(nullptr)
, mLatencyRequest(HighLatency)
, mReadPoint(0)
, mDumpFile(nullptr)
, mBytesPerFrame(0)
, mState(INITIALIZED)
, mNeedsStart(false)
, mShouldDropFrames(false)
, mPendingAudioInitTask(false)
, mLastGoodPosition(0)
{
// keep a ref in case we shut down later than nsLayoutStatics
mLatencyLog = AsyncLatencyLogger::Get(true);
}
AudioStream::~AudioStream()
@ -164,10 +153,8 @@ AudioStream::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
// Possibly add in the future:
// - mTimeStretcher
// - mLatencyLog
// - mCubebStream
amount += mInserts.ShallowSizeOfExcludingThis(aMallocSizeOf);
amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf);
return amount;
@ -319,12 +306,9 @@ WriteDumpFile(FILE* aDumpFile, AudioStream* aStream, uint32_t aFrames,
fflush(aDumpFile);
}
// NOTE: this must not block a LowLatency stream for any significant amount
// of time, or it will block the entirety of MSG
nsresult
AudioStream::Init(int32_t aNumChannels, int32_t aRate,
const dom::AudioChannel aAudioChannel,
LatencyRequest aLatencyRequest)
const dom::AudioChannel aAudioChannel)
{
mStartTime = TimeStamp::Now();
mIsFirst = CubebUtils::GetFirstStream();
@ -338,7 +322,6 @@ AudioStream::Init(int32_t aNumChannels, int32_t aRate,
mInRate = mOutRate = aRate;
mChannels = aNumChannels;
mOutChannels = (aNumChannels > 2) ? 2 : aNumChannels;
mLatencyRequest = aLatencyRequest;
mDumpFile = OpenDumpFile(this);
@ -374,108 +357,13 @@ AudioStream::Init(int32_t aNumChannels, int32_t aRate,
MOZ_ASSERT(bufferLimit % mBytesPerFrame == 0, "Must buffer complete frames");
mBuffer.SetCapacity(bufferLimit);
if (aLatencyRequest == LowLatency) {
// Don't block this thread to initialize a cubeb stream.
// When this is done, it will start callbacks from Cubeb. Those will
// cause us to move from INITIALIZED to RUNNING. Until then, we
// can't access any cubeb functions.
// Use a RefPtr to avoid leaks if Dispatch fails
mPendingAudioInitTask = true;
RefPtr<AudioInitTask> init = new AudioInitTask(this, aLatencyRequest, params);
nsresult rv = init->Dispatch();
if (NS_FAILED(rv)) {
mPendingAudioInitTask = false;
}
return rv;
}
// High latency - open synchronously
nsresult rv = OpenCubeb(params, aLatencyRequest);
NS_ENSURE_SUCCESS(rv, rv);
// See if we need to start() the stream, since we must do that from this
// thread for now (cubeb API issue)
{
MonitorAutoLock mon(mMonitor);
CheckForStart();
}
return NS_OK;
}
// On certain MacBookPro, the microphone is located near the left speaker.
// We need to pan the sound output to the right speaker if we are using the mic
// and the built-in speaker, or we will have terrible echo.
void AudioStream::PanOutputIfNeeded(bool aMicrophoneActive)
{
#ifdef XP_MACOSX
cubeb_device* device;
int rv;
char name[128];
size_t length = sizeof(name);
bool panCenter = false;
rv = sysctlbyname("hw.model", name, &length, NULL, 0);
if (rv) {
return;
}
if (!strncmp(name, "MacBookPro", 10)) {
if (cubeb_stream_get_current_device(mCubebStream.get(), &device) == CUBEB_OK) {
// Check if we are currently outputing sound on external speakers.
if (!strcmp(device->output_name, "ispk")) {
// Pan everything to the right speaker.
if (aMicrophoneActive) {
LOG(("%p Panning audio output to the right.", this));
if (cubeb_stream_set_panning(mCubebStream.get(), 1.0) != CUBEB_OK) {
NS_WARNING("Could not pan audio output to the right.");
}
} else {
panCenter = true;
}
} else {
panCenter = true;
}
if (panCenter) {
LOG(("%p Panning audio output to the center.", this));
if (cubeb_stream_set_panning(mCubebStream.get(), 0.0) != CUBEB_OK) {
NS_WARNING("Could not pan audio output to the center.");
}
}
cubeb_stream_device_destroy(mCubebStream.get(), device);
}
}
#endif
}
void AudioStream::ResetStreamIfNeeded()
{
cubeb_device * device;
// Only reset a device if a mic is active, and this is a low latency stream.
if (!mMicrophoneActive || mLatencyRequest != LowLatency) {
return;
}
if (cubeb_stream_get_current_device(mCubebStream.get(), &device) == CUBEB_OK) {
// This a microphone that goes through the headphone plug, reset the
// output to prevent echo building up.
if (strcmp(device->input_name, "emic") == 0) {
LOG(("Resetting audio output"));
Reset();
}
cubeb_stream_device_destroy(mCubebStream.get(), device);
}
}
void AudioStream::DeviceChangedCallback()
{
MonitorAutoLock mon(mMonitor);
PanOutputIfNeeded(mMicrophoneActive);
mShouldDropFrames = true;
ResetStreamIfNeeded();
return OpenCubeb(params);
}
// This code used to live inside AudioStream::Init(), but on Mac (others?)
// it has been known to take 300-800 (or even 8500) ms to execute(!)
nsresult
AudioStream::OpenCubeb(cubeb_stream_params &aParams,
LatencyRequest aLatencyRequest)
AudioStream::OpenCubeb(cubeb_stream_params &aParams)
{
cubeb* cubebContext = CubebUtils::GetCubebContext();
if (!cubebContext) {
@ -488,14 +376,7 @@ AudioStream::OpenCubeb(cubeb_stream_params &aParams,
// If the latency pref is set, use it. Otherwise, if this stream is intended
// for low latency playback, try to get the lowest latency possible.
// Otherwise, for normal streams, use 100ms.
uint32_t latency;
if (aLatencyRequest == LowLatency && !CubebUtils::CubebLatencyPrefSet()) {
if (cubeb_get_min_latency(cubebContext, aParams, &latency) != CUBEB_OK) {
latency = CubebUtils::GetCubebLatency();
}
} else {
latency = CubebUtils::GetCubebLatency();
}
uint32_t latency = CubebUtils::GetCubebLatency();
{
cubeb_stream* stream;
@ -504,9 +385,6 @@ AudioStream::OpenCubeb(cubeb_stream_params &aParams,
MonitorAutoLock mon(mMonitor);
MOZ_ASSERT(mState != SHUTDOWN);
mCubebStream.reset(stream);
// We can't cubeb_stream_start() the thread from a transient thread due to
// cubeb API requirements (init can be called from another thread, but
// not start/stop/destroy/etc)
} else {
MonitorAutoLock mon(mMonitor);
mState = ERRORED;
@ -515,9 +393,6 @@ AudioStream::OpenCubeb(cubeb_stream_params &aParams,
}
}
cubeb_stream_register_device_changed_callback(mCubebStream.get(),
AudioStream::DeviceChangedCallback_s);
mState = INITIALIZED;
if (!mStartTime.IsNull()) {
@ -531,72 +406,12 @@ AudioStream::OpenCubeb(cubeb_stream_params &aParams,
return NS_OK;
}
void
AudioStream::AudioInitTaskFinished()
{
MonitorAutoLock mon(mMonitor);
mPendingAudioInitTask = false;
mon.NotifyAll();
}
void
AudioStream::CheckForStart()
{
mMonitor.AssertCurrentThreadOwns();
if (mState == INITIALIZED) {
// Start the stream right away when low latency has been requested. This means
// that the DataCallback will feed silence to cubeb, until the first frames
// are written to this AudioStream. Also start if a start has been queued.
if (mLatencyRequest == LowLatency || mNeedsStart) {
StartUnlocked(); // mState = STARTED or ERRORED
mNeedsStart = false;
MOZ_LOG(gAudioStreamLog, LogLevel::Warning,
("Started waiting %s-latency stream",
mLatencyRequest == LowLatency ? "low" : "high"));
} else {
// high latency, not full - OR Pause() was called before we got here
MOZ_LOG(gAudioStreamLog, LogLevel::Debug,
("Not starting waiting %s-latency stream",
mLatencyRequest == LowLatency ? "low" : "high"));
}
}
}
NS_IMETHODIMP
AudioInitTask::Run()
{
MOZ_ASSERT(mThread);
if (NS_IsMainThread()) {
mThread->Shutdown(); // can't Shutdown from the thread itself, darn
// Don't null out mThread!
// See bug 999104. We must hold a ref to the thread across Dispatch()
// since the internal mThread ref could be released while processing
// the Dispatch(), and Dispatch/PutEvent itself doesn't hold a ref; it
// assumes the caller does.
return NS_OK;
}
nsresult rv = mAudioStream->OpenCubeb(mParams, mLatencyRequest);
mAudioStream->AudioInitTaskFinished();
// and now kill this thread
NS_DispatchToMainThread(this);
return rv;
}
// aTime is the time in ms the samples were inserted into MediaStreamGraph
nsresult
AudioStream::Write(const AudioDataValue* aBuf, uint32_t aFrames, TimeStamp *aTime)
AudioStream::Write(const AudioDataValue* aBuf, uint32_t aFrames)
{
MonitorAutoLock mon(mMonitor);
// See if we need to start() the stream, since we must do that from this thread
CheckForStart();
if (mShouldDropFrames) {
mBuffer.ContractTo(0);
return NS_OK;
}
if (mState == ERRORED) {
return NS_ERROR_FAILURE;
}
@ -614,22 +429,6 @@ AudioStream::Write(const AudioDataValue* aBuf, uint32_t aFrames, TimeStamp *aTim
const uint8_t* src = reinterpret_cast<const uint8_t*>(aBuf);
uint32_t bytesToCopy = FramesToBytes(aFrames);
// XXX this will need to change if we want to enable this on-the-fly!
if (MOZ_LOG_TEST(GetLatencyLog(), LogLevel::Debug)) {
// Record the position and time this data was inserted
int64_t timeMs;
if (aTime && !aTime->IsNull()) {
if (mStartTime.IsNull()) {
AsyncLatencyLogger::Get(true)->GetStartTime(mStartTime);
}
timeMs = (*aTime - mStartTime).ToMilliseconds();
} else {
timeMs = 0;
}
struct Inserts insert = { timeMs, aFrames};
mInserts.AppendElement(insert);
}
while (bytesToCopy > 0) {
uint32_t available = std::min(bytesToCopy, mBuffer.Available());
MOZ_ASSERT(available % mBytesPerFrame == 0,
@ -640,33 +439,19 @@ AudioStream::Write(const AudioDataValue* aBuf, uint32_t aFrames, TimeStamp *aTim
bytesToCopy -= available;
if (bytesToCopy > 0) {
// Careful - the CubebInit thread may not have gotten to STARTED yet
if ((mState == INITIALIZED || mState == STARTED) && mLatencyRequest == LowLatency) {
// don't ever block MediaStreamGraph low-latency streams
uint32_t remains = 0; // we presume the buffer is full
if (mBuffer.Length() > bytesToCopy) {
remains = mBuffer.Length() - bytesToCopy; // Free up just enough space
}
// account for dropping samples
MOZ_LOG(gAudioStreamLog, LogLevel::Warning, ("Stream %p dropping %u bytes (%u frames)in Write()",
this, mBuffer.Length() - remains, BytesToFrames(mBuffer.Length() - remains)));
mReadPoint += BytesToFrames(mBuffer.Length() - remains);
mBuffer.ContractTo(remains);
} else { // RUNNING or high latency
// If we are not playing, but our buffer is full, start playing to make
// room for soon-to-be-decoded data.
if (mState != STARTED && mState != RUNNING) {
MOZ_LOG(gAudioStreamLog, LogLevel::Warning, ("Starting stream %p in Write (%u waiting)",
this, bytesToCopy));
StartUnlocked();
if (mState == ERRORED) {
return NS_ERROR_FAILURE;
}
}
MOZ_LOG(gAudioStreamLog, LogLevel::Warning, ("Stream %p waiting in Write() (%u waiting)",
this, bytesToCopy));
mon.Wait();
}
// If we are not playing, but our buffer is full, start playing to make
// room for soon-to-be-decoded data.
if (mState != STARTED && mState != RUNNING) {
MOZ_LOG(gAudioStreamLog, LogLevel::Warning, ("Starting stream %p in Write (%u waiting)",
this, bytesToCopy));
StartUnlocked();
if (mState == ERRORED) {
return NS_ERROR_FAILURE;
}
}
MOZ_LOG(gAudioStreamLog, LogLevel::Warning, ("Stream %p waiting in Write() (%u waiting)",
this, bytesToCopy));
mon.Wait();
}
}
@ -692,16 +477,6 @@ AudioStream::SetVolume(double aVolume)
}
}
void
AudioStream::SetMicrophoneActive(bool aActive)
{
MonitorAutoLock mon(mMonitor);
mMicrophoneActive = aActive;
PanOutputIfNeeded(mMicrophoneActive);
}
void
AudioStream::Cancel()
{
@ -737,7 +512,6 @@ AudioStream::StartUnlocked()
{
mMonitor.AssertCurrentThreadOwns();
if (!mCubebStream) {
mNeedsStart = true;
return;
}
@ -746,8 +520,6 @@ AudioStream::StartUnlocked()
{
MonitorAutoUnlock mon(mMonitor);
r = cubeb_stream_start(mCubebStream.get());
PanOutputIfNeeded(mMicrophoneActive);
}
mState = r == CUBEB_OK ? STARTED : ERRORED;
LOG(("AudioStream: started %p, state %s", this, mState == STARTED ? "STARTED" : "ERRORED"));
@ -764,7 +536,6 @@ AudioStream::Pause()
}
if (!mCubebStream || (mState != STARTED && mState != RUNNING)) {
mNeedsStart = false;
mState = STOPPED; // which also tells async OpenCubeb not to start, just init
return;
}
@ -803,10 +574,6 @@ AudioStream::Shutdown()
MonitorAutoLock mon(mMonitor);
LOG(("AudioStream: Shutdown %p, state %d", this, mState));
while (mPendingAudioInitTask) {
mon.Wait();
}
if (mCubebStream) {
MonitorAutoUnlock mon(mMonitor);
// Force stop to put the cubeb stream in a stable state before deletion.
@ -865,17 +632,6 @@ AudioStream::GetPositionInFramesUnlocked()
return std::min<uint64_t>(mLastGoodPosition, INT64_MAX);
}
int64_t
AudioStream::GetLatencyInFrames()
{
uint32_t latency;
if (cubeb_stream_get_latency(mCubebStream.get(), &latency)) {
NS_WARNING("Could not get cubeb latency.");
return 0;
}
return static_cast<int64_t>(latency);
}
bool
AudioStream::IsPaused()
{
@ -883,26 +639,8 @@ AudioStream::IsPaused()
return mState == STOPPED;
}
void
AudioStream::GetBufferInsertTime(int64_t &aTimeMs)
{
mMonitor.AssertCurrentThreadOwns();
if (mInserts.Length() > 0) {
// Find the right block, but don't leave the array empty
while (mInserts.Length() > 1 && mReadPoint >= mInserts[0].mFrames) {
mReadPoint -= mInserts[0].mFrames;
mInserts.RemoveElementAt(0);
}
// offset for amount already read
// XXX Note: could misreport if we couldn't find a block in the right timeframe
aTimeMs = mInserts[0].mTimeMs + ((mReadPoint * 1000) / mOutRate);
} else {
aTimeMs = INT64_MAX;
}
}
long
AudioStream::GetUnprocessed(void* aBuffer, long aFrames, int64_t &aTimeMs)
AudioStream::GetUnprocessed(void* aBuffer, long aFrames)
{
mMonitor.AssertCurrentThreadOwns();
uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
@ -924,42 +662,11 @@ AudioStream::GetUnprocessed(void* aBuffer, long aFrames, int64_t &aTimeMs)
wpos += input_size[0];
memcpy(wpos, input[1], input_size[1]);
// First time block now has our first returned sample
mReadPoint += BytesToFrames(available);
GetBufferInsertTime(aTimeMs);
return BytesToFrames(available) + flushedFrames;
}
// Get unprocessed samples, and pad the beginning of the buffer with silence if
// there is not enough data.
long
AudioStream::GetUnprocessedWithSilencePadding(void* aBuffer, long aFrames, int64_t& aTimeMs)
{
mMonitor.AssertCurrentThreadOwns();
uint32_t toPopBytes = FramesToBytes(aFrames);
uint32_t available = std::min(toPopBytes, mBuffer.Length());
uint32_t silenceOffset = toPopBytes - available;
uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
memset(wpos, 0, silenceOffset);
wpos += silenceOffset;
void* input[2];
uint32_t input_size[2];
mBuffer.PopElements(available, &input[0], &input_size[0], &input[1], &input_size[1]);
memcpy(wpos, input[0], input_size[0]);
wpos += input_size[0];
memcpy(wpos, input[1], input_size[1]);
GetBufferInsertTime(aTimeMs);
return aFrames;
}
long
AudioStream::GetTimeStretched(void* aBuffer, long aFrames, int64_t &aTimeMs)
AudioStream::GetTimeStretched(void* aBuffer, long aFrames)
{
mMonitor.AssertCurrentThreadOwns();
long processedFrames = 0;
@ -985,7 +692,6 @@ AudioStream::GetTimeStretched(void* aBuffer, long aFrames, int64_t &aTimeMs)
}
mBuffer.PopElements(available, &input[0], &input_size[0],
&input[1], &input_size[1]);
mReadPoint += BytesToFrames(available);
for(uint32_t i = 0; i < 2; i++) {
mTimeStretcher->putSamples(reinterpret_cast<AudioDataValue*>(input[i]), BytesToFrames(input_size[i]));
}
@ -995,59 +701,9 @@ AudioStream::GetTimeStretched(void* aBuffer, long aFrames, int64_t &aTimeMs)
processedFrames += receivedFrames;
} while (processedFrames < aFrames && !lowOnBufferedData);
GetBufferInsertTime(aTimeMs);
return processedFrames;
}
void
AudioStream::Reset()
{
MOZ_ASSERT(mLatencyRequest == LowLatency, "We should only be reseting low latency streams");
mShouldDropFrames = true;
mNeedsStart = true;
cubeb_stream_params params;
params.rate = mInRate;
params.channels = mOutChannels;
#if defined(__ANDROID__)
#if defined(MOZ_B2G)
params.stream_type = CubebUtils::ConvertChannelToCubebType(mAudioChannel);
#else
params.stream_type = CUBEB_STREAM_TYPE_MUSIC;
#endif
if (params.stream_type == CUBEB_STREAM_TYPE_MAX) {
return;
}
#endif
if (AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_S16) {
params.format = CUBEB_SAMPLE_S16NE;
} else {
params.format = CUBEB_SAMPLE_FLOAT32NE;
}
mBytesPerFrame = sizeof(AudioDataValue) * mOutChannels;
// Size mBuffer for one second of audio. This value is arbitrary, and was
// selected based on the observed behaviour of the existing AudioStream
// implementations.
uint32_t bufferLimit = FramesToBytes(mInRate);
MOZ_ASSERT(bufferLimit % mBytesPerFrame == 0, "Must buffer complete frames");
mBuffer.Reset();
mBuffer.SetCapacity(bufferLimit);
// Don't block this thread to initialize a cubeb stream.
// When this is done, it will start callbacks from Cubeb. Those will
// cause us to move from INITIALIZED to RUNNING. Until then, we
// can't access any cubeb functions.
// Use a RefPtr to avoid leaks if Dispatch fails
RefPtr<AudioInitTask> init = new AudioInitTask(this, mLatencyRequest, params);
init->Dispatch();
}
long
AudioStream::DataCallback(void* aBuffer, long aFrames)
{
@ -1058,63 +714,26 @@ AudioStream::DataCallback(void* aBuffer, long aFrames)
AudioDataValue* output = reinterpret_cast<AudioDataValue*>(aBuffer);
uint32_t underrunFrames = 0;
uint32_t servicedFrames = 0;
int64_t insertTime;
mShouldDropFrames = false;
// NOTE: wasapi (others?) can call us back *after* stop()/Shutdown() (mState == SHUTDOWN)
// Bug 996162
// callback tells us cubeb succeeded initializing
if (mState == STARTED) {
// For low-latency streams, we want to minimize any built-up data when
// we start getting callbacks.
// Simple version - contract on first callback only.
if (mLatencyRequest == LowLatency) {
uint32_t old_len = mBuffer.Length();
available = mBuffer.ContractTo(FramesToBytes(aFrames));
TimeStamp now = TimeStamp::Now();
if (!mStartTime.IsNull()) {
int64_t timeMs = (now - mStartTime).ToMilliseconds();
MOZ_LOG(gAudioStreamLog, LogLevel::Warning,
("Stream took %lldms to start after first Write() @ %u", timeMs, mOutRate));
} else {
MOZ_LOG(gAudioStreamLog, LogLevel::Warning,
("Stream started before Write() @ %u", mOutRate));
}
if (old_len != available) {
// Note that we may have dropped samples in Write() as well!
MOZ_LOG(gAudioStreamLog, LogLevel::Warning,
("AudioStream %p dropped %u + %u initial frames @ %u", this,
mReadPoint, BytesToFrames(old_len - available), mOutRate));
mReadPoint += BytesToFrames(old_len - available);
}
}
mState = RUNNING;
}
if (available) {
// When we are playing a low latency stream, and it is the first time we are
// getting data from the buffer, we prefer to add the silence for an
// underrun at the beginning of the buffer, so the first buffer is not cut
// in half by the silence inserted to compensate for the underrun.
if (mInRate == mOutRate) {
if (mLatencyRequest == LowLatency && !mWritten) {
servicedFrames = GetUnprocessedWithSilencePadding(output, aFrames, insertTime);
} else {
servicedFrames = GetUnprocessed(output, aFrames, insertTime);
}
servicedFrames = GetUnprocessed(output, aFrames);
} else {
servicedFrames = GetTimeStretched(output, aFrames, insertTime);
servicedFrames = GetTimeStretched(output, aFrames);
}
MOZ_ASSERT(mBuffer.Length() % mBytesPerFrame == 0, "Must copy complete frames");
// Notify any blocked Write() call that more space is available in mBuffer.
mon.NotifyAll();
} else {
GetBufferInsertTime(insertTime);
}
underrunFrames = aFrames - servicedFrames;
@ -1135,21 +754,6 @@ AudioStream::DataCallback(void* aBuffer, long aFrames)
}
WriteDumpFile(mDumpFile, this, aFrames, aBuffer);
// Don't log if we're not interested or if the stream is inactive
if (MOZ_LOG_TEST(GetLatencyLog(), LogLevel::Debug) &&
mState != SHUTDOWN &&
insertTime != INT64_MAX && servicedFrames > underrunFrames) {
uint32_t latency = UINT32_MAX;
if (cubeb_stream_get_latency(mCubebStream.get(), &latency)) {
NS_WARNING("Could not get latency from cubeb.");
}
TimeStamp now = TimeStamp::Now();
mLatencyLog->Log(AsyncLatencyLogger::AudioStream, reinterpret_cast<uint64_t>(this),
insertTime, now);
mLatencyLog->Log(AsyncLatencyLogger::Cubeb, reinterpret_cast<uint64_t>(mCubebStream.get()),
(latency * 1000) / mOutRate, now);
}
return servicedFrames;
}

130
dom/media/AudioStream.h
View File

@ -10,9 +10,10 @@
#include "nsAutoPtr.h"
#include "nsCOMPtr.h"
#include "nsThreadUtils.h"
#include "Latency.h"
#include "mozilla/dom/AudioChannelBinding.h"
#include "mozilla/Monitor.h"
#include "mozilla/RefPtr.h"
#include "mozilla/TimeStamp.h"
#include "mozilla/UniquePtr.h"
#include "CubebUtils.h"
#include "soundtouch/SoundTouchFactory.h"
@ -71,7 +72,7 @@ private:
int mInRate;
// True if the we are timestretching, false if we are resampling.
bool mPreservesPitch;
// The history of frames sent to the audio engine in each Datacallback.
// The history of frames sent to the audio engine in each DataCallback.
const nsAutoPtr<FrameHistory> mFrameHistory;
};
@ -133,19 +134,6 @@ public:
mStart %= mCapacity;
}
// Throw away all but aSize bytes from the buffer. Returns new size, which
// may be less than aSize
uint32_t ContractTo(uint32_t aSize) {
MOZ_ASSERT(mBuffer && mCapacity, "Buffer not initialized.");
if (aSize >= mCount) {
return mCount;
}
mStart += (mCount - aSize);
mCount = aSize;
mStart %= mCapacity;
return mCount;
}
size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
{
size_t amount = 0;
@ -153,14 +141,6 @@ public:
return amount;
}
void Reset()
{
mBuffer = nullptr;
mCapacity = 0;
mStart = 0;
mCount = 0;
}
private:
nsAutoArrayPtr<uint8_t> mBuffer;
uint32_t mCapacity;
@ -168,8 +148,6 @@ private:
uint32_t mCount;
};
class AudioInitTask;
// Access to a single instance of this class must be synchronized by
// callers, or made from a single thread. One exception is that access to
// GetPosition, GetPositionInFrames, SetVolume, and Get{Rate,Channels},
@ -182,17 +160,11 @@ public:
NS_INLINE_DECL_THREADSAFE_REFCOUNTING(AudioStream)
AudioStream();
enum LatencyRequest {
HighLatency,
LowLatency
};
// Initialize the audio stream. aNumChannels is the number of audio
// channels (1 for mono, 2 for stereo, etc) and aRate is the sample rate
// (22050Hz, 44100Hz, etc).
nsresult Init(int32_t aNumChannels, int32_t aRate,
const dom::AudioChannel aAudioStreamChannel,
LatencyRequest aLatencyRequest);
const dom::AudioChannel aAudioStreamChannel);
// Closes the stream. All future use of the stream is an error.
void Shutdown();
@ -202,9 +174,8 @@ public:
// Write audio data to the audio hardware. aBuf is an array of AudioDataValues
// AudioDataValue of length aFrames*mChannels. If aFrames is larger
// than the result of Available(), the write will block until sufficient
// buffer space is available. aTime is the time in ms associated with the first sample
// for latency calculations
nsresult Write(const AudioDataValue* aBuf, uint32_t aFrames, TimeStamp* aTime = nullptr);
// buffer space is available.
nsresult Write(const AudioDataValue* aBuf, uint32_t aFrames);
// Return the number of audio frames that can be written without blocking.
uint32_t Available();
@ -213,12 +184,6 @@ public:
// 0 (meaning muted) to 1 (meaning full volume). Thread-safe.
void SetVolume(double aVolume);
// Informs the AudioStream that a microphone is being used by someone in the
// application.
void SetMicrophoneActive(bool aActive);
void PanOutputIfNeeded(bool aMicrophoneActive);
void ResetStreamIfNeeded();
// Block until buffered audio data has been consumed.
void Drain();
@ -271,14 +236,7 @@ protected:
int64_t GetPositionInFramesUnlocked();
private:
friend class AudioInitTask;
// So we can call it asynchronously from AudioInitTask
nsresult OpenCubeb(cubeb_stream_params &aParams,
LatencyRequest aLatencyRequest);
void AudioInitTaskFinished();
void CheckForStart();
nsresult OpenCubeb(cubeb_stream_params &aParams);
static long DataCallback_S(cubeb_stream*, void* aThis, void* aBuffer, long aFrames)
{
@ -291,23 +249,13 @@ private:
}
static void DeviceChangedCallback_s(void * aThis) {
static_cast<AudioStream*>(aThis)->DeviceChangedCallback();
}
long DataCallback(void* aBuffer, long aFrames);
void StateCallback(cubeb_state aState);
void DeviceChangedCallback();
nsresult EnsureTimeStretcherInitializedUnlocked();
// aTime is the time in ms the samples were inserted into MediaStreamGraph
long GetUnprocessed(void* aBuffer, long aFrames, int64_t &aTime);
long GetTimeStretched(void* aBuffer, long aFrames, int64_t &aTime);
long GetUnprocessedWithSilencePadding(void* aBuffer, long aFrames, int64_t &aTime);
int64_t GetLatencyInFrames();
void GetBufferInsertTime(int64_t &aTimeMs);
long GetUnprocessed(void* aBuffer, long aFrames);
long GetTimeStretched(void* aBuffer, long aFrames);
void StartUnlocked();
@ -330,22 +278,9 @@ private:
int64_t mWritten;
AudioClock mAudioClock;
soundtouch::SoundTouch* mTimeStretcher;
nsRefPtr<AsyncLatencyLogger> mLatencyLog;
// copy of Latency logger's starting time for offset calculations
// Stream start time for stream open delay telemetry.
TimeStamp mStartTime;
// Whether we are playing a low latency stream, or a normal stream.
LatencyRequest mLatencyRequest;
// Where in the current mInserts[0] block cubeb has read to
int64_t mReadPoint;
// Keep track of each inserted block of samples and the time it was inserted
// so we can estimate the clock time for a specific sample's insertion (for when
// we send data to cubeb). Blocks are aged out as needed.
struct Inserts {
int64_t mTimeMs;
int64_t mFrames;
};
nsAutoTArray<Inserts, 8> mInserts;
// Output file for dumping audio
FILE* mDumpFile;
@ -386,57 +321,12 @@ private:
};
StreamState mState;
bool mNeedsStart; // needed in case Start() is called before cubeb is open
bool mIsFirst;
// True if a microphone is active.
bool mMicrophoneActive;
// When we are in the process of changing the output device, and the callback
// is not going to be called for a little while, simply drop incoming frames.
// This is only on OSX for now, because other systems handle this gracefully.
bool mShouldDropFrames;
// True if there is a pending AudioInitTask. Shutdown() will wait until the
// pending AudioInitTask is finished.
bool mPendingAudioInitTask;
// The last good position returned by cubeb_stream_get_position(). Used to
// check if the cubeb position is going backward.
uint64_t mLastGoodPosition;
};
class AudioInitTask : public nsRunnable
{
public:
AudioInitTask(AudioStream *aStream,
AudioStream::LatencyRequest aLatencyRequest,
const cubeb_stream_params &aParams)
: mAudioStream(aStream)
, mLatencyRequest(aLatencyRequest)
, mParams(aParams)
{}
nsresult Dispatch()
{
// Can't add 'this' as the event to run, since mThread may not be set yet
nsresult rv = NS_NewNamedThread("CubebInit", getter_AddRefs(mThread));
if (NS_SUCCEEDED(rv)) {
// Note: event must not null out mThread!
rv = mThread->Dispatch(this, NS_DISPATCH_NORMAL);
}
return rv;
}
protected:
virtual ~AudioInitTask() {};
private:
NS_IMETHOD Run() override final;
RefPtr<AudioStream> mAudioStream;
AudioStream::LatencyRequest mLatencyRequest;
cubeb_stream_params mParams;
nsCOMPtr<nsIThread> mThread;
};
} // namespace mozilla
#endif

3
dom/media/mediasink/DecodedAudioDataSink.cpp
View File

@ -263,8 +263,7 @@ DecodedAudioDataSink::InitializeAudioStream()
// circumstances, so we take care to drop the decoder monitor while
// initializing.
RefPtr<AudioStream> audioStream(new AudioStream());
nsresult rv = audioStream->Init(mInfo.mChannels, mInfo.mRate,
mChannel, AudioStream::HighLatency);
nsresult rv = audioStream->Init(mInfo.mChannels, mInfo.mRate, mChannel);
if (NS_FAILED(rv)) {
audioStream->Shutdown();
return rv;