mirror of
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f012888dd8
MozReview-Commit-ID: 5eqBlPnHVk2
712 lines
21 KiB
C++
712 lines
21 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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/* vim:set ts=2 sw=2 sts=2 et cindent: */
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/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this
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* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
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#include <stdio.h>
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#include <math.h>
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#include <string.h>
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#include "mozilla/Logging.h"
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#include "prdtoa.h"
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#include "AudioStream.h"
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#include "VideoUtils.h"
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#include "mozilla/Monitor.h"
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#include "mozilla/Mutex.h"
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#include "mozilla/Sprintf.h"
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#include <algorithm>
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#include "mozilla/Telemetry.h"
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#include "CubebUtils.h"
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#include "nsPrintfCString.h"
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#include "gfxPrefs.h"
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#include "AudioConverter.h"
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namespace mozilla {
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#undef LOG
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#undef LOGW
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LazyLogModule gAudioStreamLog("AudioStream");
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// For simple logs
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#define LOG(x, ...) MOZ_LOG(gAudioStreamLog, mozilla::LogLevel::Debug, ("%p " x, this, ##__VA_ARGS__))
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#define LOGW(x, ...) MOZ_LOG(gAudioStreamLog, mozilla::LogLevel::Warning, ("%p " x, this, ##__VA_ARGS__))
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/**
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* Keep a list of frames sent to the audio engine in each DataCallback along
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* with the playback rate at the moment. Since the playback rate and number of
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* underrun frames can vary in each callback. We need to keep the whole history
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* in order to calculate the playback position of the audio engine correctly.
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*/
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class FrameHistory {
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struct Chunk {
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uint32_t servicedFrames;
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uint32_t totalFrames;
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uint32_t rate;
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};
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template <typename T>
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static T FramesToUs(uint32_t frames, int rate) {
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return static_cast<T>(frames) * USECS_PER_S / rate;
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}
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public:
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FrameHistory()
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: mBaseOffset(0), mBasePosition(0) {}
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void Append(uint32_t aServiced, uint32_t aUnderrun, uint32_t aRate) {
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/* In most case where playback rate stays the same and we don't underrun
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* frames, we are able to merge chunks to avoid lose of precision to add up
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* in compressing chunks into |mBaseOffset| and |mBasePosition|.
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*/
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if (!mChunks.IsEmpty()) {
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Chunk& c = mChunks.LastElement();
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// 2 chunks (c1 and c2) can be merged when rate is the same and
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// adjacent frames are zero. That is, underrun frames in c1 are zero
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// or serviced frames in c2 are zero.
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if (c.rate == aRate &&
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(c.servicedFrames == c.totalFrames ||
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aServiced == 0)) {
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c.servicedFrames += aServiced;
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c.totalFrames += aServiced + aUnderrun;
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return;
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}
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}
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Chunk* p = mChunks.AppendElement();
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p->servicedFrames = aServiced;
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p->totalFrames = aServiced + aUnderrun;
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p->rate = aRate;
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}
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/**
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* @param frames The playback position in frames of the audio engine.
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* @return The playback position in microseconds of the audio engine,
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* adjusted by playback rate changes and underrun frames.
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*/
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int64_t GetPosition(int64_t frames) {
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// playback position should not go backward.
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MOZ_ASSERT(frames >= mBaseOffset);
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while (true) {
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if (mChunks.IsEmpty()) {
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return mBasePosition;
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}
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const Chunk& c = mChunks[0];
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if (frames <= mBaseOffset + c.totalFrames) {
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uint32_t delta = frames - mBaseOffset;
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delta = std::min(delta, c.servicedFrames);
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return static_cast<int64_t>(mBasePosition) +
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FramesToUs<int64_t>(delta, c.rate);
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}
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// Since the playback position of the audio engine will not go backward,
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// we are able to compress chunks so that |mChunks| won't grow unlimitedly.
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// Note that we lose precision in converting integers into floats and
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// inaccuracy will accumulate over time. However, for a 24hr long,
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// sample rate = 44.1k file, the error will be less than 1 microsecond
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// after playing 24 hours. So we are fine with that.
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mBaseOffset += c.totalFrames;
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mBasePosition += FramesToUs<double>(c.servicedFrames, c.rate);
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mChunks.RemoveElementAt(0);
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}
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}
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private:
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AutoTArray<Chunk, 7> mChunks;
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int64_t mBaseOffset;
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double mBasePosition;
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};
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AudioStream::AudioStream(DataSource& aSource)
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: mMonitor("AudioStream")
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, mChannels(0)
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, mOutChannels(0)
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, mTimeStretcher(nullptr)
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, mDumpFile(nullptr)
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, mState(INITIALIZED)
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, mDataSource(aSource)
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{
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}
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AudioStream::~AudioStream()
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{
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LOG("deleted, state %d", mState);
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MOZ_ASSERT(mState == SHUTDOWN && !mCubebStream,
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"Should've called Shutdown() before deleting an AudioStream");
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if (mDumpFile) {
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fclose(mDumpFile);
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}
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if (mTimeStretcher) {
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soundtouch::destroySoundTouchObj(mTimeStretcher);
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}
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}
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size_t
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AudioStream::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
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{
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size_t amount = aMallocSizeOf(this);
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// Possibly add in the future:
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// - mTimeStretcher
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// - mCubebStream
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return amount;
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}
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nsresult AudioStream::EnsureTimeStretcherInitializedUnlocked()
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{
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mMonitor.AssertCurrentThreadOwns();
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if (!mTimeStretcher) {
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mTimeStretcher = soundtouch::createSoundTouchObj();
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mTimeStretcher->setSampleRate(mAudioClock.GetInputRate());
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mTimeStretcher->setChannels(mOutChannels);
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mTimeStretcher->setPitch(1.0);
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}
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return NS_OK;
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}
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nsresult AudioStream::SetPlaybackRate(double aPlaybackRate)
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{
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// MUST lock since the rate transposer is used from the cubeb callback,
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// and rate changes can cause the buffer to be reallocated
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MonitorAutoLock mon(mMonitor);
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NS_ASSERTION(aPlaybackRate > 0.0,
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"Can't handle negative or null playbackrate in the AudioStream.");
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// Avoid instantiating the resampler if we are not changing the playback rate.
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// GetPreservesPitch/SetPreservesPitch don't need locking before calling
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if (aPlaybackRate == mAudioClock.GetPlaybackRate()) {
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return NS_OK;
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}
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if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
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return NS_ERROR_FAILURE;
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}
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mAudioClock.SetPlaybackRate(aPlaybackRate);
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if (mAudioClock.GetPreservesPitch()) {
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mTimeStretcher->setTempo(aPlaybackRate);
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mTimeStretcher->setRate(1.0f);
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} else {
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mTimeStretcher->setTempo(1.0f);
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mTimeStretcher->setRate(aPlaybackRate);
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}
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return NS_OK;
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}
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nsresult AudioStream::SetPreservesPitch(bool aPreservesPitch)
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{
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// MUST lock since the rate transposer is used from the cubeb callback,
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// and rate changes can cause the buffer to be reallocated
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MonitorAutoLock mon(mMonitor);
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// Avoid instantiating the timestretcher instance if not needed.
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if (aPreservesPitch == mAudioClock.GetPreservesPitch()) {
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return NS_OK;
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}
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if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
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return NS_ERROR_FAILURE;
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}
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if (aPreservesPitch == true) {
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mTimeStretcher->setTempo(mAudioClock.GetPlaybackRate());
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mTimeStretcher->setRate(1.0f);
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} else {
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mTimeStretcher->setTempo(1.0f);
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mTimeStretcher->setRate(mAudioClock.GetPlaybackRate());
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}
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mAudioClock.SetPreservesPitch(aPreservesPitch);
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return NS_OK;
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}
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static void SetUint16LE(uint8_t* aDest, uint16_t aValue)
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{
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aDest[0] = aValue & 0xFF;
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aDest[1] = aValue >> 8;
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}
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static void SetUint32LE(uint8_t* aDest, uint32_t aValue)
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{
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SetUint16LE(aDest, aValue & 0xFFFF);
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SetUint16LE(aDest + 2, aValue >> 16);
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}
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static FILE*
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OpenDumpFile(uint32_t aChannels, uint32_t aRate)
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{
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/**
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* When MOZ_DUMP_AUDIO is set in the environment (to anything),
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* we'll drop a series of files in the current working directory named
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* dumped-audio-<nnn>.wav, one per AudioStream created, containing
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* the audio for the stream including any skips due to underruns.
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*/
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static Atomic<int> gDumpedAudioCount(0);
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if (!getenv("MOZ_DUMP_AUDIO"))
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return nullptr;
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char buf[100];
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SprintfLiteral(buf, "dumped-audio-%d.wav", ++gDumpedAudioCount);
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FILE* f = fopen(buf, "wb");
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if (!f)
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return nullptr;
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uint8_t header[] = {
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// RIFF header
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0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56, 0x45,
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// fmt chunk. We always write 16-bit samples.
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0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00, 0xFF, 0xFF,
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0xFF, 0xFF, 0xFF, 0xFF, 0x00, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x10, 0x00,
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// data chunk
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0x64, 0x61, 0x74, 0x61, 0xFE, 0xFF, 0xFF, 0x7F
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};
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static const int CHANNEL_OFFSET = 22;
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static const int SAMPLE_RATE_OFFSET = 24;
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static const int BLOCK_ALIGN_OFFSET = 32;
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SetUint16LE(header + CHANNEL_OFFSET, aChannels);
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SetUint32LE(header + SAMPLE_RATE_OFFSET, aRate);
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SetUint16LE(header + BLOCK_ALIGN_OFFSET, aChannels * 2);
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fwrite(header, sizeof(header), 1, f);
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return f;
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}
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template <typename T>
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typename EnableIf<IsSame<T, int16_t>::value, void>::Type
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WriteDumpFileHelper(T* aInput, size_t aSamples, FILE* aFile) {
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fwrite(aInput, sizeof(T), aSamples, aFile);
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}
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template <typename T>
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typename EnableIf<IsSame<T, float>::value, void>::Type
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WriteDumpFileHelper(T* aInput, size_t aSamples, FILE* aFile) {
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AutoTArray<uint8_t, 1024*2> buf;
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buf.SetLength(aSamples*2);
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uint8_t* output = buf.Elements();
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for (uint32_t i = 0; i < aSamples; ++i) {
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SetUint16LE(output + i*2, int16_t(aInput[i]*32767.0f));
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}
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fwrite(output, 2, aSamples, aFile);
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fflush(aFile);
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}
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static void
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WriteDumpFile(FILE* aDumpFile, AudioStream* aStream, uint32_t aFrames,
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void* aBuffer)
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{
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if (!aDumpFile)
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return;
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uint32_t samples = aStream->GetOutChannels()*aFrames;
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using SampleT = AudioSampleTraits<AUDIO_OUTPUT_FORMAT>::Type;
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WriteDumpFileHelper(reinterpret_cast<SampleT*>(aBuffer), samples, aDumpFile);
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}
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template <AudioSampleFormat N>
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struct ToCubebFormat {
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static const cubeb_sample_format value = CUBEB_SAMPLE_FLOAT32NE;
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};
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template <>
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struct ToCubebFormat<AUDIO_FORMAT_S16> {
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static const cubeb_sample_format value = CUBEB_SAMPLE_S16NE;
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};
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template <typename Function, typename... Args>
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int AudioStream::InvokeCubeb(Function aFunction, Args&&... aArgs)
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{
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MonitorAutoUnlock mon(mMonitor);
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return aFunction(mCubebStream.get(), Forward<Args>(aArgs)...);
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}
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nsresult
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AudioStream::Init(uint32_t aNumChannels, uint32_t aRate,
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const dom::AudioChannel aAudioChannel)
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{
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auto startTime = TimeStamp::Now();
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LOG("%s channels: %d, rate: %d", __FUNCTION__, aNumChannels, aRate);
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mChannels = aNumChannels;
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mOutChannels = aNumChannels;
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mDumpFile = OpenDumpFile(aNumChannels, aRate);
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cubeb_stream_params params;
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params.rate = aRate;
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params.channels = mOutChannels;
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#if defined(__ANDROID__)
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#if defined(MOZ_B2G)
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params.stream_type = CubebUtils::ConvertChannelToCubebType(aAudioChannel);
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#else
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params.stream_type = CUBEB_STREAM_TYPE_MUSIC;
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#endif
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if (params.stream_type == CUBEB_STREAM_TYPE_MAX) {
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return NS_ERROR_INVALID_ARG;
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}
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#endif
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params.format = ToCubebFormat<AUDIO_OUTPUT_FORMAT>::value;
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mAudioClock.Init(aRate);
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cubeb* cubebContext = CubebUtils::GetCubebContext();
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if (!cubebContext) {
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NS_WARNING("Can't get cubeb context!");
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CubebUtils::ReportCubebStreamInitFailure(true);
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return NS_ERROR_DOM_MEDIA_CUBEB_INITIALIZATION_ERR;
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}
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// The DecodedAudioDataSink forces mono or stereo for now.
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params.layout = params.channels == 1 ? CUBEB_LAYOUT_MONO
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: CUBEB_LAYOUT_STEREO;
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return OpenCubeb(cubebContext, params, startTime, CubebUtils::GetFirstStream());
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}
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nsresult
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AudioStream::OpenCubeb(cubeb* aContext, cubeb_stream_params& aParams,
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TimeStamp aStartTime, bool aIsFirst)
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{
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MOZ_ASSERT(aContext);
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cubeb_stream* stream = nullptr;
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/* Convert from milliseconds to frames. */
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uint32_t latency_frames =
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CubebUtils::GetCubebPlaybackLatencyInMilliseconds() * aParams.rate / 1000;
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if (cubeb_stream_init(aContext, &stream, "AudioStream",
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nullptr, nullptr, nullptr, &aParams,
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latency_frames,
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DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
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mCubebStream.reset(stream);
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CubebUtils::ReportCubebBackendUsed();
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} else {
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NS_WARNING(nsPrintfCString("AudioStream::OpenCubeb() %p failed to init cubeb", this).get());
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CubebUtils::ReportCubebStreamInitFailure(aIsFirst);
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return NS_ERROR_FAILURE;
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}
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TimeDuration timeDelta = TimeStamp::Now() - aStartTime;
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LOG("creation time %sfirst: %u ms", aIsFirst ? "" : "not ",
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(uint32_t) timeDelta.ToMilliseconds());
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Telemetry::Accumulate(aIsFirst ? Telemetry::AUDIOSTREAM_FIRST_OPEN_MS :
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Telemetry::AUDIOSTREAM_LATER_OPEN_MS, timeDelta.ToMilliseconds());
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return NS_OK;
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}
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void
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AudioStream::SetVolume(double aVolume)
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{
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MOZ_ASSERT(aVolume >= 0.0 && aVolume <= 1.0, "Invalid volume");
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if (cubeb_stream_set_volume(mCubebStream.get(), aVolume * CubebUtils::GetVolumeScale()) != CUBEB_OK) {
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NS_WARNING("Could not change volume on cubeb stream.");
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}
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}
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void
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AudioStream::Start()
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{
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MonitorAutoLock mon(mMonitor);
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MOZ_ASSERT(mState == INITIALIZED);
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mState = STARTED;
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auto r = InvokeCubeb(cubeb_stream_start);
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if (r != CUBEB_OK) {
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mState = ERRORED;
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}
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LOG("started, state %s", mState == STARTED ? "STARTED" : mState == DRAINED ? "DRAINED" : "ERRORED");
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}
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void
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AudioStream::Pause()
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{
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MonitorAutoLock mon(mMonitor);
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MOZ_ASSERT(mState != INITIALIZED, "Must be Start()ed.");
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MOZ_ASSERT(mState != STOPPED, "Already Pause()ed.");
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MOZ_ASSERT(mState != SHUTDOWN, "Already Shutdown()ed.");
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// Do nothing if we are already drained or errored.
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if (mState == DRAINED || mState == ERRORED) {
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return;
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}
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if (InvokeCubeb(cubeb_stream_stop) != CUBEB_OK) {
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mState = ERRORED;
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} else if (mState != DRAINED && mState != ERRORED) {
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// Don't transition to other states if we are already
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// drained or errored.
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mState = STOPPED;
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}
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}
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void
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AudioStream::Resume()
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{
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MonitorAutoLock mon(mMonitor);
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MOZ_ASSERT(mState != INITIALIZED, "Must be Start()ed.");
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MOZ_ASSERT(mState != STARTED, "Already Start()ed.");
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MOZ_ASSERT(mState != SHUTDOWN, "Already Shutdown()ed.");
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// Do nothing if we are already drained or errored.
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if (mState == DRAINED || mState == ERRORED) {
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return;
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}
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if (InvokeCubeb(cubeb_stream_start) != CUBEB_OK) {
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mState = ERRORED;
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} else if (mState != DRAINED && mState != ERRORED) {
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// Don't transition to other states if we are already
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// drained or errored.
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mState = STARTED;
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}
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}
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void
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AudioStream::Shutdown()
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{
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MonitorAutoLock mon(mMonitor);
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LOG("Shutdown, state %d", mState);
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if (mCubebStream) {
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MonitorAutoUnlock mon(mMonitor);
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// Force stop to put the cubeb stream in a stable state before deletion.
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cubeb_stream_stop(mCubebStream.get());
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// Must not try to shut down cubeb from within the lock! wasapi may still
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// call our callback after Pause()/stop()!?! Bug 996162
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mCubebStream.reset();
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}
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mState = SHUTDOWN;
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}
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int64_t
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AudioStream::GetPosition()
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{
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|
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
|