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-rw-r--r--dom/media/AudioStream.cpp707
1 files changed, 707 insertions, 0 deletions
diff --git a/dom/media/AudioStream.cpp b/dom/media/AudioStream.cpp
new file mode 100644
index 000000000..82ae3ee86
--- /dev/null
+++ b/dom/media/AudioStream.cpp
@@ -0,0 +1,707 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+#include <stdio.h>
+#include <math.h>
+#include <string.h>
+#include "mozilla/Logging.h"
+#include "prdtoa.h"
+#include "AudioStream.h"
+#include "VideoUtils.h"
+#include "mozilla/Monitor.h"
+#include "mozilla/Mutex.h"
+#include "mozilla/Sprintf.h"
+#include <algorithm>
+#include "mozilla/Telemetry.h"
+#include "CubebUtils.h"
+#include "nsPrintfCString.h"
+#include "gfxPrefs.h"
+#include "AudioConverter.h"
+
+namespace mozilla {
+
+#undef LOG
+#undef LOGW
+
+LazyLogModule gAudioStreamLog("AudioStream");
+// For simple logs
+#define LOG(x, ...) MOZ_LOG(gAudioStreamLog, mozilla::LogLevel::Debug, ("%p " x, this, ##__VA_ARGS__))
+#define LOGW(x, ...) MOZ_LOG(gAudioStreamLog, mozilla::LogLevel::Warning, ("%p " x, this, ##__VA_ARGS__))
+
+/**
+ * Keep a list of frames sent to the audio engine in each DataCallback along
+ * with the playback rate at the moment. Since the playback rate and number of
+ * underrun frames can vary in each callback. We need to keep the whole history
+ * in order to calculate the playback position of the audio engine correctly.
+ */
+class FrameHistory {
+ struct Chunk {
+ uint32_t servicedFrames;
+ uint32_t totalFrames;
+ uint32_t rate;
+ };
+
+ template <typename T>
+ static T FramesToUs(uint32_t frames, int rate) {
+ return static_cast<T>(frames) * USECS_PER_S / rate;
+ }
+public:
+ FrameHistory()
+ : mBaseOffset(0), mBasePosition(0) {}
+
+ void Append(uint32_t aServiced, uint32_t aUnderrun, uint32_t aRate) {
+ /* In most case where playback rate stays the same and we don't underrun
+ * frames, we are able to merge chunks to avoid lose of precision to add up
+ * in compressing chunks into |mBaseOffset| and |mBasePosition|.
+ */
+ if (!mChunks.IsEmpty()) {
+ Chunk& c = mChunks.LastElement();
+ // 2 chunks (c1 and c2) can be merged when rate is the same and
+ // adjacent frames are zero. That is, underrun frames in c1 are zero
+ // or serviced frames in c2 are zero.
+ if (c.rate == aRate &&
+ (c.servicedFrames == c.totalFrames ||
+ aServiced == 0)) {
+ c.servicedFrames += aServiced;
+ c.totalFrames += aServiced + aUnderrun;
+ return;
+ }
+ }
+ Chunk* p = mChunks.AppendElement();
+ p->servicedFrames = aServiced;
+ p->totalFrames = aServiced + aUnderrun;
+ p->rate = aRate;
+ }
+
+ /**
+ * @param frames The playback position in frames of the audio engine.
+ * @return The playback position in microseconds of the audio engine,
+ * adjusted by playback rate changes and underrun frames.
+ */
+ int64_t GetPosition(int64_t frames) {
+ // playback position should not go backward.
+ MOZ_ASSERT(frames >= mBaseOffset);
+ while (true) {
+ if (mChunks.IsEmpty()) {
+ return mBasePosition;
+ }
+ const Chunk& c = mChunks[0];
+ if (frames <= mBaseOffset + c.totalFrames) {
+ uint32_t delta = frames - mBaseOffset;
+ delta = std::min(delta, c.servicedFrames);
+ return static_cast<int64_t>(mBasePosition) +
+ FramesToUs<int64_t>(delta, c.rate);
+ }
+ // Since the playback position of the audio engine will not go backward,
+ // we are able to compress chunks so that |mChunks| won't grow unlimitedly.
+ // Note that we lose precision in converting integers into floats and
+ // inaccuracy will accumulate over time. However, for a 24hr long,
+ // sample rate = 44.1k file, the error will be less than 1 microsecond
+ // after playing 24 hours. So we are fine with that.
+ mBaseOffset += c.totalFrames;
+ mBasePosition += FramesToUs<double>(c.servicedFrames, c.rate);
+ mChunks.RemoveElementAt(0);
+ }
+ }
+private:
+ AutoTArray<Chunk, 7> mChunks;
+ int64_t mBaseOffset;
+ double mBasePosition;
+};
+
+AudioStream::AudioStream(DataSource& aSource)
+ : mMonitor("AudioStream")
+ , mChannels(0)
+ , mOutChannels(0)
+ , mTimeStretcher(nullptr)
+ , mDumpFile(nullptr)
+ , mState(INITIALIZED)
+ , mDataSource(aSource)
+{
+}
+
+AudioStream::~AudioStream()
+{
+ LOG("deleted, state %d", mState);
+ MOZ_ASSERT(mState == SHUTDOWN && !mCubebStream,
+ "Should've called Shutdown() before deleting an AudioStream");
+ if (mDumpFile) {
+ fclose(mDumpFile);
+ }
+ if (mTimeStretcher) {
+ soundtouch::destroySoundTouchObj(mTimeStretcher);
+ }
+}
+
+size_t
+AudioStream::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+ size_t amount = aMallocSizeOf(this);
+
+ // Possibly add in the future:
+ // - mTimeStretcher
+ // - mCubebStream
+
+ return amount;
+}
+
+nsresult AudioStream::EnsureTimeStretcherInitializedUnlocked()
+{
+ mMonitor.AssertCurrentThreadOwns();
+ if (!mTimeStretcher) {
+ mTimeStretcher = soundtouch::createSoundTouchObj();
+ mTimeStretcher->setSampleRate(mAudioClock.GetInputRate());
+ mTimeStretcher->setChannels(mOutChannels);
+ mTimeStretcher->setPitch(1.0);
+ }
+ return NS_OK;
+}
+
+nsresult AudioStream::SetPlaybackRate(double aPlaybackRate)
+{
+ // MUST lock since the rate transposer is used from the cubeb callback,
+ // and rate changes can cause the buffer to be reallocated
+ MonitorAutoLock mon(mMonitor);
+
+ NS_ASSERTION(aPlaybackRate > 0.0,
+ "Can't handle negative or null playbackrate in the AudioStream.");
+ // Avoid instantiating the resampler if we are not changing the playback rate.
+ // GetPreservesPitch/SetPreservesPitch don't need locking before calling
+ if (aPlaybackRate == mAudioClock.GetPlaybackRate()) {
+ return NS_OK;
+ }
+
+ if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
+ return NS_ERROR_FAILURE;
+ }
+
+ mAudioClock.SetPlaybackRate(aPlaybackRate);
+
+ if (mAudioClock.GetPreservesPitch()) {
+ mTimeStretcher->setTempo(aPlaybackRate);
+ mTimeStretcher->setRate(1.0f);
+ } else {
+ mTimeStretcher->setTempo(1.0f);
+ mTimeStretcher->setRate(aPlaybackRate);
+ }
+ return NS_OK;
+}
+
+nsresult AudioStream::SetPreservesPitch(bool aPreservesPitch)
+{
+ // MUST lock since the rate transposer is used from the cubeb callback,
+ // and rate changes can cause the buffer to be reallocated
+ MonitorAutoLock mon(mMonitor);
+
+ // Avoid instantiating the timestretcher instance if not needed.
+ if (aPreservesPitch == mAudioClock.GetPreservesPitch()) {
+ return NS_OK;
+ }
+
+ if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
+ return NS_ERROR_FAILURE;
+ }
+
+ if (aPreservesPitch == true) {
+ mTimeStretcher->setTempo(mAudioClock.GetPlaybackRate());
+ mTimeStretcher->setRate(1.0f);
+ } else {
+ mTimeStretcher->setTempo(1.0f);
+ mTimeStretcher->setRate(mAudioClock.GetPlaybackRate());
+ }
+
+ mAudioClock.SetPreservesPitch(aPreservesPitch);
+
+ return NS_OK;
+}
+
+static void SetUint16LE(uint8_t* aDest, uint16_t aValue)
+{
+ aDest[0] = aValue & 0xFF;
+ aDest[1] = aValue >> 8;
+}
+
+static void SetUint32LE(uint8_t* aDest, uint32_t aValue)
+{
+ SetUint16LE(aDest, aValue & 0xFFFF);
+ SetUint16LE(aDest + 2, aValue >> 16);
+}
+
+static FILE*
+OpenDumpFile(uint32_t aChannels, uint32_t aRate)
+{
+ /**
+ * When MOZ_DUMP_AUDIO is set in the environment (to anything),
+ * we'll drop a series of files in the current working directory named
+ * dumped-audio-<nnn>.wav, one per AudioStream created, containing
+ * the audio for the stream including any skips due to underruns.
+ */
+ static Atomic<int> gDumpedAudioCount(0);
+
+ if (!getenv("MOZ_DUMP_AUDIO"))
+ return nullptr;
+ char buf[100];
+ SprintfLiteral(buf, "dumped-audio-%d.wav", ++gDumpedAudioCount);
+ FILE* f = fopen(buf, "wb");
+ if (!f)
+ return nullptr;
+
+ uint8_t header[] = {
+ // RIFF header
+ 0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56, 0x45,
+ // fmt chunk. We always write 16-bit samples.
+ 0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00, 0xFF, 0xFF,
+ 0xFF, 0xFF, 0xFF, 0xFF, 0x00, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x10, 0x00,
+ // data chunk
+ 0x64, 0x61, 0x74, 0x61, 0xFE, 0xFF, 0xFF, 0x7F
+ };
+ static const int CHANNEL_OFFSET = 22;
+ static const int SAMPLE_RATE_OFFSET = 24;
+ static const int BLOCK_ALIGN_OFFSET = 32;
+ SetUint16LE(header + CHANNEL_OFFSET, aChannels);
+ SetUint32LE(header + SAMPLE_RATE_OFFSET, aRate);
+ SetUint16LE(header + BLOCK_ALIGN_OFFSET, aChannels * 2);
+ fwrite(header, sizeof(header), 1, f);
+
+ return f;
+}
+
+template <typename T>
+typename EnableIf<IsSame<T, int16_t>::value, void>::Type
+WriteDumpFileHelper(T* aInput, size_t aSamples, FILE* aFile) {
+ fwrite(aInput, sizeof(T), aSamples, aFile);
+}
+
+template <typename T>
+typename EnableIf<IsSame<T, float>::value, void>::Type
+WriteDumpFileHelper(T* aInput, size_t aSamples, FILE* aFile) {
+ AutoTArray<uint8_t, 1024*2> buf;
+ buf.SetLength(aSamples*2);
+ uint8_t* output = buf.Elements();
+ for (uint32_t i = 0; i < aSamples; ++i) {
+ SetUint16LE(output + i*2, int16_t(aInput[i]*32767.0f));
+ }
+ fwrite(output, 2, aSamples, aFile);
+ fflush(aFile);
+}
+
+static void
+WriteDumpFile(FILE* aDumpFile, AudioStream* aStream, uint32_t aFrames,
+ void* aBuffer)
+{
+ if (!aDumpFile)
+ return;
+
+ uint32_t samples = aStream->GetOutChannels()*aFrames;
+
+ using SampleT = AudioSampleTraits<AUDIO_OUTPUT_FORMAT>::Type;
+ WriteDumpFileHelper(reinterpret_cast<SampleT*>(aBuffer), samples, aDumpFile);
+}
+
+template <AudioSampleFormat N>
+struct ToCubebFormat {
+ static const cubeb_sample_format value = CUBEB_SAMPLE_FLOAT32NE;
+};
+
+template <>
+struct ToCubebFormat<AUDIO_FORMAT_S16> {
+ static const cubeb_sample_format value = CUBEB_SAMPLE_S16NE;
+};
+
+template <typename Function, typename... Args>
+int AudioStream::InvokeCubeb(Function aFunction, Args&&... aArgs)
+{
+ MonitorAutoUnlock mon(mMonitor);
+ return aFunction(mCubebStream.get(), Forward<Args>(aArgs)...);
+}
+
+nsresult
+AudioStream::Init(uint32_t aNumChannels, uint32_t aRate,
+ const dom::AudioChannel aAudioChannel)
+{
+ auto startTime = TimeStamp::Now();
+
+ LOG("%s channels: %d, rate: %d", __FUNCTION__, aNumChannels, aRate);
+ mChannels = aNumChannels;
+ mOutChannels = aNumChannels;
+
+ mDumpFile = OpenDumpFile(aNumChannels, aRate);
+
+ cubeb_stream_params params;
+ params.rate = aRate;
+ params.channels = mOutChannels;
+#if defined(__ANDROID__)
+#if defined(MOZ_B2G)
+ params.stream_type = CubebUtils::ConvertChannelToCubebType(aAudioChannel);
+#else
+ params.stream_type = CUBEB_STREAM_TYPE_MUSIC;
+#endif
+
+ if (params.stream_type == CUBEB_STREAM_TYPE_MAX) {
+ return NS_ERROR_INVALID_ARG;
+ }
+#endif
+
+ params.format = ToCubebFormat<AUDIO_OUTPUT_FORMAT>::value;
+ mAudioClock.Init(aRate);
+
+ cubeb* cubebContext = CubebUtils::GetCubebContext();
+ if (!cubebContext) {
+ NS_WARNING("Can't get cubeb context!");
+ CubebUtils::ReportCubebStreamInitFailure(true);
+ return NS_ERROR_DOM_MEDIA_CUBEB_INITIALIZATION_ERR;
+ }
+
+ return OpenCubeb(cubebContext, params, startTime, CubebUtils::GetFirstStream());
+}
+
+nsresult
+AudioStream::OpenCubeb(cubeb* aContext, cubeb_stream_params& aParams,
+ TimeStamp aStartTime, bool aIsFirst)
+{
+ MOZ_ASSERT(aContext);
+
+ cubeb_stream* stream = nullptr;
+ /* Convert from milliseconds to frames. */
+ uint32_t latency_frames =
+ CubebUtils::GetCubebPlaybackLatencyInMilliseconds() * aParams.rate / 1000;
+ if (cubeb_stream_init(aContext, &stream, "AudioStream",
+ nullptr, nullptr, nullptr, &aParams,
+ latency_frames,
+ DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
+ mCubebStream.reset(stream);
+ CubebUtils::ReportCubebBackendUsed();
+ } else {
+ NS_WARNING(nsPrintfCString("AudioStream::OpenCubeb() %p failed to init cubeb", this).get());
+ CubebUtils::ReportCubebStreamInitFailure(aIsFirst);
+ return NS_ERROR_FAILURE;
+ }
+
+ TimeDuration timeDelta = TimeStamp::Now() - aStartTime;
+ LOG("creation time %sfirst: %u ms", aIsFirst ? "" : "not ",
+ (uint32_t) timeDelta.ToMilliseconds());
+ Telemetry::Accumulate(aIsFirst ? Telemetry::AUDIOSTREAM_FIRST_OPEN_MS :
+ Telemetry::AUDIOSTREAM_LATER_OPEN_MS, timeDelta.ToMilliseconds());
+
+ return NS_OK;
+}
+
+void
+AudioStream::SetVolume(double aVolume)
+{
+ MOZ_ASSERT(aVolume >= 0.0 && aVolume <= 1.0, "Invalid volume");
+
+ if (cubeb_stream_set_volume(mCubebStream.get(), aVolume * CubebUtils::GetVolumeScale()) != CUBEB_OK) {
+ NS_WARNING("Could not change volume on cubeb stream.");
+ }
+}
+
+void
+AudioStream::Start()
+{
+ MonitorAutoLock mon(mMonitor);
+ MOZ_ASSERT(mState == INITIALIZED);
+ mState = STARTED;
+ auto r = InvokeCubeb(cubeb_stream_start);
+ if (r != CUBEB_OK) {
+ mState = ERRORED;
+ }
+ LOG("started, state %s", mState == STARTED ? "STARTED" : mState == DRAINED ? "DRAINED" : "ERRORED");
+}
+
+void
+AudioStream::Pause()
+{
+ MonitorAutoLock mon(mMonitor);
+ MOZ_ASSERT(mState != INITIALIZED, "Must be Start()ed.");
+ MOZ_ASSERT(mState != STOPPED, "Already Pause()ed.");
+ MOZ_ASSERT(mState != SHUTDOWN, "Already Shutdown()ed.");
+
+ // Do nothing if we are already drained or errored.
+ if (mState == DRAINED || mState == ERRORED) {
+ return;
+ }
+
+ if (InvokeCubeb(cubeb_stream_stop) != CUBEB_OK) {
+ mState = ERRORED;
+ } else if (mState != DRAINED && mState != ERRORED) {
+ // Don't transition to other states if we are already
+ // drained or errored.
+ mState = STOPPED;
+ }
+}
+
+void
+AudioStream::Resume()
+{
+ MonitorAutoLock mon(mMonitor);
+ MOZ_ASSERT(mState != INITIALIZED, "Must be Start()ed.");
+ MOZ_ASSERT(mState != STARTED, "Already Start()ed.");
+ MOZ_ASSERT(mState != SHUTDOWN, "Already Shutdown()ed.");
+
+ // Do nothing if we are already drained or errored.
+ if (mState == DRAINED || mState == ERRORED) {
+ return;
+ }
+
+ if (InvokeCubeb(cubeb_stream_start) != CUBEB_OK) {
+ mState = ERRORED;
+ } else if (mState != DRAINED && mState != ERRORED) {
+ // Don't transition to other states if we are already
+ // drained or errored.
+ mState = STARTED;
+ }
+}
+
+void
+AudioStream::Shutdown()
+{
+ MonitorAutoLock mon(mMonitor);
+ LOG("Shutdown, state %d", mState);
+
+ if (mCubebStream) {
+ MonitorAutoUnlock mon(mMonitor);
+ // Force stop to put the cubeb stream in a stable state before deletion.
+ cubeb_stream_stop(mCubebStream.get());
+ // Must not try to shut down cubeb from within the lock! wasapi may still
+ // call our callback after Pause()/stop()!?! Bug 996162
+ mCubebStream.reset();
+ }
+
+ mState = SHUTDOWN;
+}
+
+int64_t
+AudioStream::GetPosition()
+{
+ MonitorAutoLock mon(mMonitor);
+ int64_t frames = GetPositionInFramesUnlocked();
+ return frames >= 0 ? mAudioClock.GetPosition(frames) : -1;
+}
+
+int64_t
+AudioStream::GetPositionInFrames()
+{
+ MonitorAutoLock mon(mMonitor);
+ int64_t frames = GetPositionInFramesUnlocked();
+ return frames >= 0 ? mAudioClock.GetPositionInFrames(frames) : -1;
+}
+
+int64_t
+AudioStream::GetPositionInFramesUnlocked()
+{
+ mMonitor.AssertCurrentThreadOwns();
+
+ if (mState == ERRORED) {
+ return -1;
+ }
+
+ uint64_t position = 0;
+ if (InvokeCubeb(cubeb_stream_get_position, &position) != CUBEB_OK) {
+ return -1;
+ }
+ return std::min<uint64_t>(position, INT64_MAX);
+}
+
+bool
+AudioStream::IsValidAudioFormat(Chunk* aChunk)
+{
+ if (aChunk->Rate() != mAudioClock.GetInputRate()) {
+ LOGW("mismatched sample %u, mInRate=%u", aChunk->Rate(), mAudioClock.GetInputRate());
+ return false;
+ }
+
+ if (aChunk->Channels() > 8) {
+ return false;
+ }
+
+ return true;
+}
+
+void
+AudioStream::GetUnprocessed(AudioBufferWriter& aWriter)
+{
+ mMonitor.AssertCurrentThreadOwns();
+
+ // Flush the timestretcher pipeline, if we were playing using a playback rate
+ // other than 1.0.
+ if (mTimeStretcher && mTimeStretcher->numSamples()) {
+ auto timeStretcher = mTimeStretcher;
+ aWriter.Write([timeStretcher] (AudioDataValue* aPtr, uint32_t aFrames) {
+ return timeStretcher->receiveSamples(aPtr, aFrames);
+ }, aWriter.Available());
+
+ // TODO: There might be still unprocessed samples in the stretcher.
+ // We should either remove or flush them so they won't be in the output
+ // next time we switch a playback rate other than 1.0.
+ NS_WARNING_ASSERTION(
+ mTimeStretcher->numUnprocessedSamples() == 0, "no samples");
+ }
+
+ while (aWriter.Available() > 0) {
+ UniquePtr<Chunk> c = mDataSource.PopFrames(aWriter.Available());
+ if (c->Frames() == 0) {
+ break;
+ }
+ MOZ_ASSERT(c->Frames() <= aWriter.Available());
+ if (IsValidAudioFormat(c.get())) {
+ aWriter.Write(c->Data(), c->Frames());
+ } else {
+ // Write silence if invalid format.
+ aWriter.WriteZeros(c->Frames());
+ }
+ }
+}
+
+void
+AudioStream::GetTimeStretched(AudioBufferWriter& aWriter)
+{
+ mMonitor.AssertCurrentThreadOwns();
+
+ // We need to call the non-locking version, because we already have the lock.
+ if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
+ return;
+ }
+
+ uint32_t toPopFrames =
+ ceil(aWriter.Available() * mAudioClock.GetPlaybackRate());
+
+ while (mTimeStretcher->numSamples() < aWriter.Available()) {
+ UniquePtr<Chunk> c = mDataSource.PopFrames(toPopFrames);
+ if (c->Frames() == 0) {
+ break;
+ }
+ MOZ_ASSERT(c->Frames() <= toPopFrames);
+ if (IsValidAudioFormat(c.get())) {
+ mTimeStretcher->putSamples(c->Data(), c->Frames());
+ } else {
+ // Write silence if invalid format.
+ AutoTArray<AudioDataValue, 1000> buf;
+ buf.SetLength(mOutChannels * c->Frames());
+ memset(buf.Elements(), 0, buf.Length() * sizeof(AudioDataValue));
+ mTimeStretcher->putSamples(buf.Elements(), c->Frames());
+ }
+ }
+
+ auto timeStretcher = mTimeStretcher;
+ aWriter.Write([timeStretcher] (AudioDataValue* aPtr, uint32_t aFrames) {
+ return timeStretcher->receiveSamples(aPtr, aFrames);
+ }, aWriter.Available());
+}
+
+long
+AudioStream::DataCallback(void* aBuffer, long aFrames)
+{
+ MonitorAutoLock mon(mMonitor);
+ MOZ_ASSERT(mState != SHUTDOWN, "No data callback after shutdown");
+
+ auto writer = AudioBufferWriter(
+ reinterpret_cast<AudioDataValue*>(aBuffer), mOutChannels, aFrames);
+
+ if (!strcmp(cubeb_get_backend_id(CubebUtils::GetCubebContext()), "winmm")) {
+ // Don't consume audio data until Start() is called.
+ // Expected only with cubeb winmm backend.
+ if (mState == INITIALIZED) {
+ NS_WARNING("data callback fires before cubeb_stream_start() is called");
+ mAudioClock.UpdateFrameHistory(0, aFrames);
+ return writer.WriteZeros(aFrames);
+ }
+ } else {
+ MOZ_ASSERT(mState != INITIALIZED);
+ }
+
+ // NOTE: wasapi (others?) can call us back *after* stop()/Shutdown() (mState == SHUTDOWN)
+ // Bug 996162
+
+ if (mAudioClock.GetInputRate() == mAudioClock.GetOutputRate()) {
+ GetUnprocessed(writer);
+ } else {
+ GetTimeStretched(writer);
+ }
+
+ // Always send audible frames first, and silent frames later.
+ // Otherwise it will break the assumption of FrameHistory.
+ if (!mDataSource.Ended()) {
+ mAudioClock.UpdateFrameHistory(aFrames - writer.Available(), writer.Available());
+ if (writer.Available() > 0) {
+ LOGW("lost %d frames", writer.Available());
+ writer.WriteZeros(writer.Available());
+ }
+ } else {
+ // No more new data in the data source. Don't send silent frames so the
+ // cubeb stream can start draining.
+ mAudioClock.UpdateFrameHistory(aFrames - writer.Available(), 0);
+ }
+
+ WriteDumpFile(mDumpFile, this, aFrames, aBuffer);
+
+ return aFrames - writer.Available();
+}
+
+void
+AudioStream::StateCallback(cubeb_state aState)
+{
+ MonitorAutoLock mon(mMonitor);
+ MOZ_ASSERT(mState != SHUTDOWN, "No state callback after shutdown");
+ LOG("StateCallback, mState=%d cubeb_state=%d", mState, aState);
+ if (aState == CUBEB_STATE_DRAINED) {
+ mState = DRAINED;
+ mDataSource.Drained();
+ } else if (aState == CUBEB_STATE_ERROR) {
+ LOG("StateCallback() state %d cubeb error", mState);
+ mState = ERRORED;
+ }
+}
+
+AudioClock::AudioClock()
+: mOutRate(0),
+ mInRate(0),
+ mPreservesPitch(true),
+ mFrameHistory(new FrameHistory())
+{}
+
+void AudioClock::Init(uint32_t aRate)
+{
+ mOutRate = aRate;
+ mInRate = aRate;
+}
+
+void AudioClock::UpdateFrameHistory(uint32_t aServiced, uint32_t aUnderrun)
+{
+ mFrameHistory->Append(aServiced, aUnderrun, mOutRate);
+}
+
+int64_t AudioClock::GetPositionInFrames(int64_t aFrames) const
+{
+ CheckedInt64 v = UsecsToFrames(GetPosition(aFrames), mInRate);
+ return v.isValid() ? v.value() : -1;
+}
+
+int64_t AudioClock::GetPosition(int64_t frames) const
+{
+ return mFrameHistory->GetPosition(frames);
+}
+
+void AudioClock::SetPlaybackRate(double aPlaybackRate)
+{
+ mOutRate = static_cast<uint32_t>(mInRate / aPlaybackRate);
+}
+
+double AudioClock::GetPlaybackRate() const
+{
+ return static_cast<double>(mInRate) / mOutRate;
+}
+
+void AudioClock::SetPreservesPitch(bool aPreservesPitch)
+{
+ mPreservesPitch = aPreservesPitch;
+}
+
+bool AudioClock::GetPreservesPitch() const
+{
+ return mPreservesPitch;
+}
+
+} // namespace mozilla