mirror of
https://gitlab.winehq.org/wine/wine-gecko.git
synced 2024-09-13 09:24:08 -07:00
1228 lines
37 KiB
C++
1228 lines
37 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 "prlog.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 <algorithm>
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#include "mozilla/Telemetry.h"
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#include "soundtouch/SoundTouch.h"
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#include "Latency.h"
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#include "CubebUtils.h"
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#include "nsPrintfCString.h"
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#ifdef XP_MACOSX
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#include <sys/sysctl.h>
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#endif
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namespace mozilla {
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#ifdef LOG
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#undef LOG
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#endif
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#ifdef PR_LOGGING
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PRLogModuleInfo* gAudioStreamLog = nullptr;
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// For simple logs
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#define LOG(x) PR_LOG(gAudioStreamLog, PR_LOG_DEBUG, x)
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#else
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#define LOG(x)
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#endif
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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 int gDumpedAudioCount = 0;
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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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int 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, int 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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nsAutoTArray<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()
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: mMonitor("AudioStream")
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, mInRate(0)
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, mOutRate(0)
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, mChannels(0)
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, mOutChannels(0)
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, mWritten(0)
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, mAudioClock(MOZ_THIS_IN_INITIALIZER_LIST())
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, mLatencyRequest(HighLatency)
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, mReadPoint(0)
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, mDumpFile(nullptr)
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, mBytesPerFrame(0)
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, mState(INITIALIZED)
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, mNeedsStart(false)
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, mShouldDropFrames(false)
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, mPendingAudioInitTask(false)
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, mLastGoodPosition(0)
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{
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// keep a ref in case we shut down later than nsLayoutStatics
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mLatencyLog = AsyncLatencyLogger::Get(true);
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}
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AudioStream::~AudioStream()
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{
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LOG(("AudioStream: delete %p, state %d", this, 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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}
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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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// - mLatencyLog
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// - mCubebStream
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amount += mInserts.SizeOfExcludingThis(aMallocSizeOf);
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amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf);
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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 = new soundtouch::SoundTouch();
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mTimeStretcher->setSampleRate(mInRate);
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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.SetPlaybackRateUnlocked(aPlaybackRate);
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mOutRate = mInRate / 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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int64_t AudioStream::GetWritten()
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{
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MonitorAutoLock mon(mMonitor);
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return mWritten;
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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(AudioStream* aStream)
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{
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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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sprintf(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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++gDumpedAudioCount;
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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, aStream->GetChannels());
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SetUint32LE(header + SAMPLE_RATE_OFFSET, aStream->GetRate());
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SetUint16LE(header + BLOCK_ALIGN_OFFSET, aStream->GetChannels()*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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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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if (AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_S16) {
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fwrite(aBuffer, 2, samples, aDumpFile);
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return;
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}
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NS_ASSERTION(AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_FLOAT32, "bad format");
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nsAutoTArray<uint8_t, 1024*2> buf;
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buf.SetLength(samples*2);
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float* input = static_cast<float*>(aBuffer);
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uint8_t* output = buf.Elements();
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for (uint32_t i = 0; i < samples; ++i) {
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SetUint16LE(output + i*2, int16_t(input[i]*32767.0f));
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}
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fwrite(output, 2, samples, aDumpFile);
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fflush(aDumpFile);
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}
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// NOTE: this must not block a LowLatency stream for any significant amount
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// of time, or it will block the entirety of MSG
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nsresult
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AudioStream::Init(int32_t aNumChannels, int32_t aRate,
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const dom::AudioChannel aAudioChannel,
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LatencyRequest aLatencyRequest)
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{
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mStartTime = TimeStamp::Now();
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mIsFirst = CubebUtils::GetFirstStream();
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if (!CubebUtils::GetCubebContext() || aNumChannels < 0 || aRate < 0) {
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return NS_ERROR_FAILURE;
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}
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PR_LOG(gAudioStreamLog, PR_LOG_DEBUG,
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("%s channels: %d, rate: %d for %p", __FUNCTION__, aNumChannels, aRate, this));
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mInRate = mOutRate = aRate;
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mChannels = aNumChannels;
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mOutChannels = (aNumChannels > 2) ? 2 : aNumChannels;
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mLatencyRequest = aLatencyRequest;
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mDumpFile = OpenDumpFile(this);
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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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mAudioChannel = aAudioChannel;
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params.stream_type = CubebUtils::ConvertChannelToCubebType(aAudioChannel);
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#else
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mAudioChannel = dom::AudioChannel::Content;
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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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if (AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_S16) {
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params.format = CUBEB_SAMPLE_S16NE;
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} else {
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params.format = CUBEB_SAMPLE_FLOAT32NE;
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}
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mBytesPerFrame = sizeof(AudioDataValue) * mOutChannels;
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mAudioClock.Init();
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// Size mBuffer for one second of audio. This value is arbitrary, and was
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// selected based on the observed behaviour of the existing AudioStream
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// implementations.
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uint32_t bufferLimit = FramesToBytes(aRate);
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NS_ABORT_IF_FALSE(bufferLimit % mBytesPerFrame == 0, "Must buffer complete frames");
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mBuffer.SetCapacity(bufferLimit);
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if (aLatencyRequest == LowLatency) {
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// Don't block this thread to initialize a cubeb stream.
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// When this is done, it will start callbacks from Cubeb. Those will
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// cause us to move from INITIALIZED to RUNNING. Until then, we
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// can't access any cubeb functions.
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// Use a RefPtr to avoid leaks if Dispatch fails
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mPendingAudioInitTask = true;
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RefPtr<AudioInitTask> init = new AudioInitTask(this, aLatencyRequest, params);
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nsresult rv = init->Dispatch();
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if (NS_FAILED(rv)) {
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mPendingAudioInitTask = false;
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}
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return rv;
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}
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// High latency - open synchronously
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nsresult rv = OpenCubeb(params, aLatencyRequest);
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NS_ENSURE_SUCCESS(rv, rv);
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// See if we need to start() the stream, since we must do that from this
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// thread for now (cubeb API issue)
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{
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MonitorAutoLock mon(mMonitor);
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CheckForStart();
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}
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return NS_OK;
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}
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// On certain MacBookPro, the microphone is located near the left speaker.
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// We need to pan the sound output to the right speaker if we are using the mic
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// and the built-in speaker, or we will have terrible echo.
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void AudioStream::PanOutputIfNeeded(bool aMicrophoneActive)
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{
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#ifdef XP_MACOSX
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cubeb_device* device;
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int rv;
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char name[128];
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size_t length = sizeof(name);
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bool panCenter = false;
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rv = sysctlbyname("hw.model", name, &length, NULL, 0);
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if (rv) {
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return;
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}
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if (!strncmp(name, "MacBookPro", 10)) {
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if (cubeb_stream_get_current_device(mCubebStream.get(), &device) == CUBEB_OK) {
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// Check if we are currently outputing sound on external speakers.
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if (!strcmp(device->output_name, "ispk")) {
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// Pan everything to the right speaker.
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if (aMicrophoneActive) {
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LOG(("%p Panning audio output to the right.", this));
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if (cubeb_stream_set_panning(mCubebStream.get(), 1.0) != CUBEB_OK) {
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NS_WARNING("Could not pan audio output to the right.");
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}
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} else {
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panCenter = true;
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}
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} else {
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panCenter = true;
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}
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if (panCenter) {
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LOG(("%p Panning audio output to the center.", this));
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if (cubeb_stream_set_panning(mCubebStream.get(), 0.0) != CUBEB_OK) {
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NS_WARNING("Could not pan audio output to the center.");
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}
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}
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cubeb_stream_device_destroy(mCubebStream.get(), device);
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}
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}
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#endif
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}
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void AudioStream::ResetStreamIfNeeded()
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{
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cubeb_device * device;
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// Only reset a device if a mic is active, and this is a low latency stream.
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if (!mMicrophoneActive || mLatencyRequest != LowLatency) {
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return;
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}
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if (cubeb_stream_get_current_device(mCubebStream.get(), &device) == CUBEB_OK) {
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// This a microphone that goes through the headphone plug, reset the
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// output to prevent echo building up.
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if (strcmp(device->input_name, "emic") == 0) {
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LOG(("Resetting audio output"));
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Reset();
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}
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cubeb_stream_device_destroy(mCubebStream.get(), device);
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}
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}
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void AudioStream::DeviceChangedCallback()
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{
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MonitorAutoLock mon(mMonitor);
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PanOutputIfNeeded(mMicrophoneActive);
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mShouldDropFrames = true;
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ResetStreamIfNeeded();
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}
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|
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// This code used to live inside AudioStream::Init(), but on Mac (others?)
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// it has been known to take 300-800 (or even 8500) ms to execute(!)
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nsresult
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AudioStream::OpenCubeb(cubeb_stream_params &aParams,
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LatencyRequest aLatencyRequest)
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{
|
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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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MonitorAutoLock mon(mMonitor);
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mState = AudioStream::ERRORED;
|
|
return NS_ERROR_FAILURE;
|
|
}
|
|
|
|
// 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();
|
|
}
|
|
|
|
{
|
|
cubeb_stream* stream;
|
|
if (cubeb_stream_init(cubebContext, &stream, "AudioStream", aParams,
|
|
latency, DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
|
|
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;
|
|
NS_WARNING(nsPrintfCString("AudioStream::OpenCubeb() %p failed to init cubeb", this).get());
|
|
return NS_ERROR_FAILURE;
|
|
}
|
|
}
|
|
|
|
cubeb_stream_register_device_changed_callback(mCubebStream.get(),
|
|
AudioStream::DeviceChangedCallback_s);
|
|
|
|
mState = INITIALIZED;
|
|
|
|
if (!mStartTime.IsNull()) {
|
|
TimeDuration timeDelta = TimeStamp::Now() - mStartTime;
|
|
LOG(("AudioStream creation time %sfirst: %u ms", mIsFirst ? "" : "not ",
|
|
(uint32_t) timeDelta.ToMilliseconds()));
|
|
Telemetry::Accumulate(mIsFirst ? Telemetry::AUDIOSTREAM_FIRST_OPEN_MS :
|
|
Telemetry::AUDIOSTREAM_LATER_OPEN_MS, timeDelta.ToMilliseconds());
|
|
}
|
|
|
|
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;
|
|
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
|
|
("Started waiting %s-latency stream",
|
|
mLatencyRequest == LowLatency ? "low" : "high"));
|
|
} else {
|
|
// high latency, not full - OR Pause() was called before we got here
|
|
PR_LOG(gAudioStreamLog, PR_LOG_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)
|
|
{
|
|
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;
|
|
}
|
|
NS_ASSERTION(mState == INITIALIZED || mState == STARTED || mState == RUNNING,
|
|
"Stream write in unexpected state.");
|
|
|
|
// Downmix to Stereo.
|
|
if (mChannels > 2 && mChannels <= 8) {
|
|
DownmixAudioToStereo(const_cast<AudioDataValue*> (aBuf), mChannels, aFrames);
|
|
}
|
|
else if (mChannels > 8) {
|
|
return NS_ERROR_FAILURE;
|
|
}
|
|
|
|
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 (PR_LOG_TEST(GetLatencyLog(), PR_LOG_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());
|
|
NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0,
|
|
"Must copy complete frames.");
|
|
|
|
mBuffer.AppendElements(src, available);
|
|
src += available;
|
|
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
|
|
PR_LOG(gAudioStreamLog, PR_LOG_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) {
|
|
PR_LOG(gAudioStreamLog, PR_LOG_WARNING, ("Starting stream %p in Write (%u waiting)",
|
|
this, bytesToCopy));
|
|
StartUnlocked();
|
|
if (mState == ERRORED) {
|
|
return NS_ERROR_FAILURE;
|
|
}
|
|
}
|
|
PR_LOG(gAudioStreamLog, PR_LOG_WARNING, ("Stream %p waiting in Write() (%u waiting)",
|
|
this, bytesToCopy));
|
|
mon.Wait();
|
|
}
|
|
}
|
|
}
|
|
|
|
mWritten += aFrames;
|
|
return NS_OK;
|
|
}
|
|
|
|
uint32_t
|
|
AudioStream::Available()
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
NS_ABORT_IF_FALSE(mBuffer.Length() % mBytesPerFrame == 0, "Buffer invariant violated.");
|
|
return BytesToFrames(mBuffer.Available());
|
|
}
|
|
|
|
void
|
|
AudioStream::SetVolume(double aVolume)
|
|
{
|
|
NS_ABORT_IF_FALSE(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::SetMicrophoneActive(bool aActive)
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
|
|
mMicrophoneActive = aActive;
|
|
|
|
PanOutputIfNeeded(mMicrophoneActive);
|
|
}
|
|
|
|
void
|
|
AudioStream::Cancel()
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
mState = ERRORED;
|
|
mon.NotifyAll();
|
|
}
|
|
|
|
void
|
|
AudioStream::Drain()
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
LOG(("AudioStream::Drain() for %p, state %d, avail %u", this, mState, mBuffer.Available()));
|
|
if (mState != STARTED && mState != RUNNING) {
|
|
NS_ASSERTION(mState == ERRORED || mBuffer.Available() == 0, "Draining without full buffer of unplayed audio");
|
|
return;
|
|
}
|
|
mState = DRAINING;
|
|
while (mState == DRAINING) {
|
|
mon.Wait();
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioStream::Start()
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
StartUnlocked();
|
|
}
|
|
|
|
void
|
|
AudioStream::StartUnlocked()
|
|
{
|
|
mMonitor.AssertCurrentThreadOwns();
|
|
if (!mCubebStream) {
|
|
mNeedsStart = true;
|
|
return;
|
|
}
|
|
|
|
if (mState == INITIALIZED) {
|
|
int r;
|
|
{
|
|
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"));
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioStream::Pause()
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
|
|
if (mState == ERRORED) {
|
|
return;
|
|
}
|
|
|
|
if (!mCubebStream || (mState != STARTED && mState != RUNNING)) {
|
|
mNeedsStart = false;
|
|
mState = STOPPED; // which also tells async OpenCubeb not to start, just init
|
|
return;
|
|
}
|
|
|
|
int r;
|
|
{
|
|
MonitorAutoUnlock mon(mMonitor);
|
|
r = cubeb_stream_stop(mCubebStream.get());
|
|
}
|
|
if (mState != ERRORED && r == CUBEB_OK) {
|
|
mState = STOPPED;
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioStream::Resume()
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
if (!mCubebStream || mState != STOPPED) {
|
|
return;
|
|
}
|
|
|
|
int r;
|
|
{
|
|
MonitorAutoUnlock mon(mMonitor);
|
|
r = cubeb_stream_start(mCubebStream.get());
|
|
}
|
|
if (mState != ERRORED && r == CUBEB_OK) {
|
|
mState = STARTED;
|
|
}
|
|
}
|
|
|
|
void
|
|
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.
|
|
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);
|
|
return mAudioClock.GetPositionUnlocked();
|
|
}
|
|
|
|
// This function is miscompiled by PGO with MSVC 2010. See bug 768333.
|
|
#ifdef _MSC_VER
|
|
#pragma optimize("", off)
|
|
#endif
|
|
int64_t
|
|
AudioStream::GetPositionInFrames()
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
return mAudioClock.GetPositionInFrames();
|
|
}
|
|
#ifdef _MSC_VER
|
|
#pragma optimize("", on)
|
|
#endif
|
|
|
|
int64_t
|
|
AudioStream::GetPositionInFramesUnlocked()
|
|
{
|
|
mMonitor.AssertCurrentThreadOwns();
|
|
|
|
if (!mCubebStream || mState == ERRORED) {
|
|
return -1;
|
|
}
|
|
|
|
uint64_t position = 0;
|
|
{
|
|
MonitorAutoUnlock mon(mMonitor);
|
|
if (cubeb_stream_get_position(mCubebStream.get(), &position) != CUBEB_OK) {
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
MOZ_ASSERT(position >= mLastGoodPosition, "cubeb position shouldn't go backward");
|
|
// This error handling/recovery keeps us in good shape in release build.
|
|
if (position >= mLastGoodPosition) {
|
|
mLastGoodPosition = position;
|
|
}
|
|
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()
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
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)
|
|
{
|
|
mMonitor.AssertCurrentThreadOwns();
|
|
uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
|
|
|
|
// Flush the timestretcher pipeline, if we were playing using a playback rate
|
|
// other than 1.0.
|
|
uint32_t flushedFrames = 0;
|
|
if (mTimeStretcher && mTimeStretcher->numSamples()) {
|
|
flushedFrames = mTimeStretcher->receiveSamples(reinterpret_cast<AudioDataValue*>(wpos), aFrames);
|
|
wpos += FramesToBytes(flushedFrames);
|
|
}
|
|
uint32_t toPopBytes = FramesToBytes(aFrames - flushedFrames);
|
|
uint32_t available = std::min(toPopBytes, mBuffer.Length());
|
|
|
|
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]);
|
|
|
|
// 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)
|
|
{
|
|
mMonitor.AssertCurrentThreadOwns();
|
|
long processedFrames = 0;
|
|
|
|
// We need to call the non-locking version, because we already have the lock.
|
|
if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
|
|
return 0;
|
|
}
|
|
|
|
uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
|
|
double playbackRate = static_cast<double>(mInRate) / mOutRate;
|
|
uint32_t toPopBytes = FramesToBytes(ceil(aFrames * playbackRate));
|
|
uint32_t available = 0;
|
|
bool lowOnBufferedData = false;
|
|
do {
|
|
// Check if we already have enough data in the time stretcher pipeline.
|
|
if (mTimeStretcher->numSamples() <= static_cast<uint32_t>(aFrames)) {
|
|
void* input[2];
|
|
uint32_t input_size[2];
|
|
available = std::min(mBuffer.Length(), toPopBytes);
|
|
if (available != toPopBytes) {
|
|
lowOnBufferedData = true;
|
|
}
|
|
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]));
|
|
}
|
|
}
|
|
uint32_t receivedFrames = mTimeStretcher->receiveSamples(reinterpret_cast<AudioDataValue*>(wpos), aFrames - processedFrames);
|
|
wpos += FramesToBytes(receivedFrames);
|
|
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);
|
|
NS_ABORT_IF_FALSE(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)
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
MOZ_ASSERT(mState != SHUTDOWN, "No data callback after shutdown");
|
|
uint32_t available = std::min(static_cast<uint32_t>(FramesToBytes(aFrames)), mBuffer.Length());
|
|
NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0, "Must copy complete frames");
|
|
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) {
|
|
#ifdef PR_LOGGING
|
|
uint32_t old_len = mBuffer.Length();
|
|
#endif
|
|
available = mBuffer.ContractTo(FramesToBytes(aFrames));
|
|
#ifdef PR_LOGGING
|
|
TimeStamp now = TimeStamp::Now();
|
|
if (!mStartTime.IsNull()) {
|
|
int64_t timeMs = (now - mStartTime).ToMilliseconds();
|
|
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
|
|
("Stream took %lldms to start after first Write() @ %u", timeMs, mOutRate));
|
|
} else {
|
|
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
|
|
("Stream started before Write() @ %u", mOutRate));
|
|
}
|
|
|
|
if (old_len != available) {
|
|
// Note that we may have dropped samples in Write() as well!
|
|
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
|
|
("AudioStream %p dropped %u + %u initial frames @ %u", this,
|
|
mReadPoint, BytesToFrames(old_len - available), mOutRate));
|
|
mReadPoint += BytesToFrames(old_len - available);
|
|
}
|
|
#endif
|
|
}
|
|
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);
|
|
}
|
|
} else {
|
|
servicedFrames = GetTimeStretched(output, aFrames, insertTime);
|
|
}
|
|
|
|
NS_ABORT_IF_FALSE(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;
|
|
|
|
// Always send audible frames first, and silent frames later.
|
|
// Otherwise it will break the assumption of FrameHistory.
|
|
if (mState != DRAINING) {
|
|
mAudioClock.UpdateFrameHistory(servicedFrames, underrunFrames);
|
|
uint8_t* rpos = static_cast<uint8_t*>(aBuffer) + FramesToBytes(aFrames - underrunFrames);
|
|
memset(rpos, 0, FramesToBytes(underrunFrames));
|
|
if (underrunFrames) {
|
|
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
|
|
("AudioStream %p lost %d frames", this, underrunFrames));
|
|
}
|
|
servicedFrames += underrunFrames;
|
|
} else {
|
|
mAudioClock.UpdateFrameHistory(servicedFrames, 0);
|
|
}
|
|
|
|
WriteDumpFile(mDumpFile, this, aFrames, aBuffer);
|
|
// Don't log if we're not interested or if the stream is inactive
|
|
if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_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;
|
|
}
|
|
|
|
void
|
|
AudioStream::StateCallback(cubeb_state aState)
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
MOZ_ASSERT(mState != SHUTDOWN, "No state callback after shutdown");
|
|
LOG(("AudioStream: StateCallback %p, mState=%d cubeb_state=%d", this, mState, aState));
|
|
if (aState == CUBEB_STATE_DRAINED) {
|
|
mState = DRAINED;
|
|
} else if (aState == CUBEB_STATE_ERROR) {
|
|
LOG(("AudioStream::StateCallback() state %d cubeb error", mState));
|
|
mState = ERRORED;
|
|
}
|
|
mon.NotifyAll();
|
|
}
|
|
|
|
AudioClock::AudioClock(AudioStream* aStream)
|
|
:mAudioStream(aStream),
|
|
mOutRate(0),
|
|
mInRate(0),
|
|
mPreservesPitch(true),
|
|
mFrameHistory(new FrameHistory())
|
|
{}
|
|
|
|
void AudioClock::Init()
|
|
{
|
|
mOutRate = mAudioStream->GetRate();
|
|
mInRate = mAudioStream->GetRate();
|
|
}
|
|
|
|
void AudioClock::UpdateFrameHistory(uint32_t aServiced, uint32_t aUnderrun)
|
|
{
|
|
mFrameHistory->Append(aServiced, aUnderrun, mOutRate);
|
|
}
|
|
|
|
int64_t AudioClock::GetPositionUnlocked() const
|
|
{
|
|
// GetPositionInFramesUnlocked() asserts it owns the monitor
|
|
int64_t frames = mAudioStream->GetPositionInFramesUnlocked();
|
|
NS_ASSERTION(frames < 0 || (mInRate != 0 && mOutRate != 0), "AudioClock not initialized.");
|
|
return frames >= 0 ? mFrameHistory->GetPosition(frames) : -1;
|
|
}
|
|
|
|
int64_t AudioClock::GetPositionInFrames() const
|
|
{
|
|
return (GetPositionUnlocked() * mInRate) / USECS_PER_S;
|
|
}
|
|
|
|
void AudioClock::SetPlaybackRateUnlocked(double aPlaybackRate)
|
|
{
|
|
mOutRate = static_cast<int>(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
|