mirror of
https://github.com/mozilla/gecko-dev.git
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968 lines
27 KiB
C++
968 lines
27 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 "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/Preferences.h"
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#include "soundtouch/SoundTouch.h"
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#include "Latency.h"
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namespace mozilla {
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#ifdef PR_LOGGING
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PRLogModuleInfo* gAudioStreamLog = nullptr;
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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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#define PREF_VOLUME_SCALE "media.volume_scale"
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#define PREF_CUBEB_LATENCY "media.cubeb_latency_ms"
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static const uint32_t CUBEB_NORMAL_LATENCY_MS = 100;
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StaticMutex AudioStream::sMutex;
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cubeb* AudioStream::sCubebContext;
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uint32_t AudioStream::sPreferredSampleRate;
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double AudioStream::sVolumeScale;
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uint32_t AudioStream::sCubebLatency;
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bool AudioStream::sCubebLatencyPrefSet;
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/*static*/ void AudioStream::PrefChanged(const char* aPref, void* aClosure)
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{
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if (strcmp(aPref, PREF_VOLUME_SCALE) == 0) {
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nsAdoptingString value = Preferences::GetString(aPref);
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StaticMutexAutoLock lock(sMutex);
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if (value.IsEmpty()) {
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sVolumeScale = 1.0;
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} else {
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NS_ConvertUTF16toUTF8 utf8(value);
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sVolumeScale = std::max<double>(0, PR_strtod(utf8.get(), nullptr));
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}
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} else if (strcmp(aPref, PREF_CUBEB_LATENCY) == 0) {
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// Arbitrary default stream latency of 100ms. The higher this
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// value, the longer stream volume changes will take to become
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// audible.
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sCubebLatencyPrefSet = Preferences::HasUserValue(aPref);
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uint32_t value = Preferences::GetUint(aPref, CUBEB_NORMAL_LATENCY_MS);
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StaticMutexAutoLock lock(sMutex);
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sCubebLatency = std::min<uint32_t>(std::max<uint32_t>(value, 1), 1000);
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}
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}
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/*static*/ double AudioStream::GetVolumeScale()
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{
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StaticMutexAutoLock lock(sMutex);
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return sVolumeScale;
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}
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/*static*/ cubeb* AudioStream::GetCubebContext()
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{
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StaticMutexAutoLock lock(sMutex);
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return GetCubebContextUnlocked();
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}
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/*static*/ void AudioStream::InitPreferredSampleRate()
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{
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StaticMutexAutoLock lock(sMutex);
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if (sPreferredSampleRate == 0 &&
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cubeb_get_preferred_sample_rate(GetCubebContextUnlocked(),
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&sPreferredSampleRate) != CUBEB_OK) {
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sPreferredSampleRate = 44100;
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}
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}
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/*static*/ cubeb* AudioStream::GetCubebContextUnlocked()
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{
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sMutex.AssertCurrentThreadOwns();
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if (sCubebContext ||
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cubeb_init(&sCubebContext, "AudioStream") == CUBEB_OK) {
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return sCubebContext;
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}
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NS_WARNING("cubeb_init failed");
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return nullptr;
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}
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/*static*/ uint32_t AudioStream::GetCubebLatency()
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{
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StaticMutexAutoLock lock(sMutex);
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return sCubebLatency;
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}
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/*static*/ bool AudioStream::CubebLatencyPrefSet()
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{
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StaticMutexAutoLock lock(sMutex);
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return sCubebLatencyPrefSet;
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}
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#if defined(__ANDROID__) && defined(MOZ_B2G)
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static cubeb_stream_type ConvertChannelToCubebType(dom::AudioChannelType aType)
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{
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switch(aType) {
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case dom::AUDIO_CHANNEL_NORMAL:
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return CUBEB_STREAM_TYPE_SYSTEM;
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case dom::AUDIO_CHANNEL_CONTENT:
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return CUBEB_STREAM_TYPE_MUSIC;
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case dom::AUDIO_CHANNEL_NOTIFICATION:
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return CUBEB_STREAM_TYPE_NOTIFICATION;
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case dom::AUDIO_CHANNEL_ALARM:
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return CUBEB_STREAM_TYPE_ALARM;
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case dom::AUDIO_CHANNEL_TELEPHONY:
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return CUBEB_STREAM_TYPE_VOICE_CALL;
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case dom::AUDIO_CHANNEL_RINGER:
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return CUBEB_STREAM_TYPE_RING;
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// Currently Android openSLES library doesn't support FORCE_AUDIBLE yet.
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case dom::AUDIO_CHANNEL_PUBLICNOTIFICATION:
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default:
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NS_ERROR("The value of AudioChannelType is invalid");
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return CUBEB_STREAM_TYPE_MAX;
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}
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}
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#endif
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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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, mLostFrames(0)
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, mDumpFile(nullptr)
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, mVolume(1.0)
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, mBytesPerFrame(0)
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, mState(INITIALIZED)
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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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Shutdown();
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if (mDumpFile) {
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fclose(mDumpFile);
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}
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}
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/*static*/ void AudioStream::InitLibrary()
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{
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#ifdef PR_LOGGING
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gAudioStreamLog = PR_NewLogModule("AudioStream");
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#endif
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PrefChanged(PREF_VOLUME_SCALE, nullptr);
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Preferences::RegisterCallback(PrefChanged, PREF_VOLUME_SCALE);
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PrefChanged(PREF_CUBEB_LATENCY, nullptr);
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Preferences::RegisterCallback(PrefChanged, PREF_CUBEB_LATENCY);
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}
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/*static*/ void AudioStream::ShutdownLibrary()
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{
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Preferences::UnregisterCallback(PrefChanged, PREF_VOLUME_SCALE);
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Preferences::UnregisterCallback(PrefChanged, PREF_CUBEB_LATENCY);
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StaticMutexAutoLock lock(sMutex);
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if (sCubebContext) {
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cubeb_destroy(sCubebContext);
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sCubebContext = nullptr;
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}
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}
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nsresult AudioStream::EnsureTimeStretcherInitialized()
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{
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MonitorAutoLock mon(mMonitor);
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return EnsureTimeStretcherInitializedUnlocked();
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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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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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if (aPlaybackRate == mAudioClock.GetPlaybackRate()) {
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return NS_OK;
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}
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if (EnsureTimeStretcherInitialized() != 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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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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// 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 (EnsureTimeStretcherInitialized() != 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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return mWritten;
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}
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/*static*/ int AudioStream::MaxNumberOfChannels()
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{
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cubeb* cubebContext = GetCubebContext();
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uint32_t maxNumberOfChannels;
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if (cubebContext &&
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cubeb_get_max_channel_count(cubebContext,
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&maxNumberOfChannels) == CUBEB_OK) {
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return static_cast<int>(maxNumberOfChannels);
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}
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return 0;
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}
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/*static*/ int AudioStream::PreferredSampleRate()
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{
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MOZ_ASSERT(sPreferredSampleRate,
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"sPreferredSampleRate has not been initialized!");
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return sPreferredSampleRate;
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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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nsresult
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AudioStream::Init(int32_t aNumChannels, int32_t aRate,
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const dom::AudioChannelType aAudioChannelType,
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LatencyRequest aLatencyRequest)
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{
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cubeb* cubebContext = GetCubebContext();
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if (!cubebContext || 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", __FUNCTION__, aNumChannels, aRate));
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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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params.stream_type = ConvertChannelToCubebType(aAudioChannelType);
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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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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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// If the latency pref is set, use it. Otherwise, if this stream is intended
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// for low latency playback, try to get the lowest latency possible.
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// Otherwise, for normal streams, use 100ms.
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uint32_t latency;
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if (aLatencyRequest == LowLatency && !CubebLatencyPrefSet()) {
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if (cubeb_get_min_latency(cubebContext, params, &latency) != CUBEB_OK) {
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latency = GetCubebLatency();
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}
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} else {
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latency = GetCubebLatency();
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}
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{
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cubeb_stream* stream;
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if (cubeb_stream_init(cubebContext, &stream, "AudioStream", params,
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latency, DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
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mCubebStream.own(stream);
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}
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}
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if (!mCubebStream) {
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return NS_ERROR_FAILURE;
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}
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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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// Start the stream right away when low latency has been requested. This means
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// that the DataCallback will feed silence to cubeb, until the first frames
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// are writtent to this AudioStream.
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if (mLatencyRequest == LowLatency) {
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Start();
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}
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return NS_OK;
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}
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void
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AudioStream::Shutdown()
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{
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if (mState == STARTED) {
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Pause();
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}
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if (mCubebStream) {
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mCubebStream.reset();
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}
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}
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// aTime is the time in ms the samples were inserted into MediaStreamGraph
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nsresult
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AudioStream::Write(const AudioDataValue* aBuf, uint32_t aFrames, TimeStamp *aTime)
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{
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MonitorAutoLock mon(mMonitor);
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if (!mCubebStream || mState == ERRORED) {
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return NS_ERROR_FAILURE;
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}
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NS_ASSERTION(mState == INITIALIZED || mState == STARTED,
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"Stream write in unexpected state.");
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// Downmix to Stereo.
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if (mChannels > 2 && mChannels <= 8) {
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DownmixAudioToStereo(const_cast<AudioDataValue*> (aBuf), mChannels, aFrames);
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}
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else if (mChannels > 8) {
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return NS_ERROR_FAILURE;
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}
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const uint8_t* src = reinterpret_cast<const uint8_t*>(aBuf);
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uint32_t bytesToCopy = FramesToBytes(aFrames);
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// XXX this will need to change if we want to enable this on-the-fly!
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if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG)) {
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// Record the position and time this data was inserted
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int64_t timeMs;
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if (aTime && !aTime->IsNull()) {
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if (mStartTime.IsNull()) {
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AsyncLatencyLogger::Get(true)->GetStartTime(mStartTime);
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}
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timeMs = (*aTime - mStartTime).ToMilliseconds();
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} else {
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timeMs = 0;
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}
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struct Inserts insert = { timeMs, aFrames};
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mInserts.AppendElement(insert);
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}
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while (bytesToCopy > 0) {
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uint32_t available = std::min(bytesToCopy, mBuffer.Available());
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NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0,
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"Must copy complete frames.");
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mBuffer.AppendElements(src, available);
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src += available;
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bytesToCopy -= available;
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if (bytesToCopy > 0) {
|
|
// If we are not playing, but our buffer is full, start playing to make
|
|
// room for soon-to-be-decoded data.
|
|
if (mState != STARTED) {
|
|
StartUnlocked();
|
|
if (mState != STARTED) {
|
|
return NS_ERROR_FAILURE;
|
|
}
|
|
}
|
|
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)
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
NS_ABORT_IF_FALSE(aVolume >= 0.0 && aVolume <= 1.0, "Invalid volume");
|
|
mVolume = aVolume;
|
|
}
|
|
|
|
void
|
|
AudioStream::Drain()
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
if (mState != STARTED) {
|
|
NS_ASSERTION(mBuffer.Available() == 0, "Draining with unplayed audio");
|
|
return;
|
|
}
|
|
mState = DRAINING;
|
|
while (mState == DRAINING) {
|
|
mon.Wait();
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioStream::Start()
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
StartUnlocked();
|
|
}
|
|
|
|
void
|
|
AudioStream::StartUnlocked()
|
|
{
|
|
mMonitor.AssertCurrentThreadOwns();
|
|
if (!mCubebStream || mState != INITIALIZED) {
|
|
return;
|
|
}
|
|
if (mState != STARTED) {
|
|
int r;
|
|
{
|
|
MonitorAutoUnlock mon(mMonitor);
|
|
r = cubeb_stream_start(mCubebStream);
|
|
}
|
|
if (mState != ERRORED) {
|
|
mState = r == CUBEB_OK ? STARTED : ERRORED;
|
|
}
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioStream::Pause()
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
if (!mCubebStream || mState != STARTED) {
|
|
return;
|
|
}
|
|
|
|
int r;
|
|
{
|
|
MonitorAutoUnlock mon(mMonitor);
|
|
r = cubeb_stream_stop(mCubebStream);
|
|
}
|
|
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);
|
|
}
|
|
if (mState != ERRORED && r == CUBEB_OK) {
|
|
mState = STARTED;
|
|
}
|
|
}
|
|
|
|
int64_t
|
|
AudioStream::GetPosition()
|
|
{
|
|
return mAudioClock.GetPosition();
|
|
}
|
|
|
|
// This function is miscompiled by PGO with MSVC 2010. See bug 768333.
|
|
#ifdef _MSC_VER
|
|
#pragma optimize("", off)
|
|
#endif
|
|
int64_t
|
|
AudioStream::GetPositionInFrames()
|
|
{
|
|
return mAudioClock.GetPositionInFrames();
|
|
}
|
|
#ifdef _MSC_VER
|
|
#pragma optimize("", on)
|
|
#endif
|
|
|
|
int64_t
|
|
AudioStream::GetPositionInFramesInternal()
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
return GetPositionInFramesUnlocked();
|
|
}
|
|
|
|
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, &position) != CUBEB_OK) {
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
// Adjust the reported position by the number of silent frames written
|
|
// during stream underruns.
|
|
uint64_t adjustedPosition = 0;
|
|
if (position >= mLostFrames) {
|
|
adjustedPosition = position - mLostFrames;
|
|
}
|
|
return std::min<uint64_t>(adjustedPosition, INT64_MAX);
|
|
}
|
|
|
|
int64_t
|
|
AudioStream::GetLatencyInFrames()
|
|
{
|
|
uint32_t latency;
|
|
if (cubeb_stream_get_latency(mCubebStream, &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)
|
|
{
|
|
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)
|
|
{
|
|
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)
|
|
{
|
|
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)
|
|
{
|
|
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;
|
|
}
|
|
|
|
long
|
|
AudioStream::DataCallback(void* aBuffer, long aFrames)
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
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;
|
|
|
|
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);
|
|
}
|
|
float scaled_volume = float(GetVolumeScale() * mVolume);
|
|
|
|
ScaleAudioSamples(output, aFrames * mOutChannels, scaled_volume);
|
|
|
|
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;
|
|
|
|
if (mState != DRAINING) {
|
|
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));
|
|
}
|
|
mLostFrames += underrunFrames;
|
|
servicedFrames += underrunFrames;
|
|
}
|
|
|
|
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) &&
|
|
insertTime != INT64_MAX && servicedFrames > underrunFrames) {
|
|
uint32_t latency = UINT32_MAX;
|
|
if (cubeb_stream_get_latency(mCubebStream, &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);
|
|
}
|
|
|
|
mAudioClock.UpdateWritePosition(servicedFrames);
|
|
return servicedFrames;
|
|
}
|
|
|
|
void
|
|
AudioStream::StateCallback(cubeb_state aState)
|
|
{
|
|
MonitorAutoLock mon(mMonitor);
|
|
if (aState == CUBEB_STATE_DRAINED) {
|
|
mState = DRAINED;
|
|
} else if (aState == CUBEB_STATE_ERROR) {
|
|
mState = ERRORED;
|
|
}
|
|
mon.NotifyAll();
|
|
}
|
|
|
|
AudioClock::AudioClock(AudioStream* aStream)
|
|
:mAudioStream(aStream),
|
|
mOldOutRate(0),
|
|
mBasePosition(0),
|
|
mBaseOffset(0),
|
|
mOldBaseOffset(0),
|
|
mOldBasePosition(0),
|
|
mPlaybackRateChangeOffset(0),
|
|
mPreviousPosition(0),
|
|
mWritten(0),
|
|
mOutRate(0),
|
|
mInRate(0),
|
|
mPreservesPitch(true),
|
|
mCompensatingLatency(false)
|
|
{}
|
|
|
|
void AudioClock::Init()
|
|
{
|
|
mOutRate = mAudioStream->GetRate();
|
|
mInRate = mAudioStream->GetRate();
|
|
mOldOutRate = mOutRate;
|
|
}
|
|
|
|
void AudioClock::UpdateWritePosition(uint32_t aCount)
|
|
{
|
|
mWritten += aCount;
|
|
}
|
|
|
|
uint64_t AudioClock::GetPosition()
|
|
{
|
|
int64_t position = mAudioStream->GetPositionInFramesInternal();
|
|
int64_t diffOffset;
|
|
NS_ASSERTION(position < 0 || (mInRate != 0 && mOutRate != 0), "AudioClock not initialized.");
|
|
if (position >= 0) {
|
|
if (position < mPlaybackRateChangeOffset) {
|
|
// See if we are still playing frames pushed with the old playback rate in
|
|
// the backend. If we are, use the old output rate to compute the
|
|
// position.
|
|
mCompensatingLatency = true;
|
|
diffOffset = position - mOldBaseOffset;
|
|
position = static_cast<uint64_t>(mOldBasePosition +
|
|
static_cast<float>(USECS_PER_S * diffOffset) / mOldOutRate);
|
|
mPreviousPosition = position;
|
|
return position;
|
|
}
|
|
|
|
if (mCompensatingLatency) {
|
|
diffOffset = position - mPlaybackRateChangeOffset;
|
|
mCompensatingLatency = false;
|
|
mBasePosition = mPreviousPosition;
|
|
} else {
|
|
diffOffset = position - mPlaybackRateChangeOffset;
|
|
}
|
|
position = static_cast<uint64_t>(mBasePosition +
|
|
(static_cast<float>(USECS_PER_S * diffOffset) / mOutRate));
|
|
return position;
|
|
}
|
|
return UINT64_MAX;
|
|
}
|
|
|
|
uint64_t AudioClock::GetPositionInFrames()
|
|
{
|
|
return (GetPosition() * mOutRate) / USECS_PER_S;
|
|
}
|
|
|
|
void AudioClock::SetPlaybackRate(double aPlaybackRate)
|
|
{
|
|
int64_t position = mAudioStream->GetPositionInFramesInternal();
|
|
if (position > mPlaybackRateChangeOffset) {
|
|
mOldBasePosition = mBasePosition;
|
|
mBasePosition = GetPosition();
|
|
mOldBaseOffset = mPlaybackRateChangeOffset;
|
|
mBaseOffset = position;
|
|
mPlaybackRateChangeOffset = mWritten;
|
|
mOldOutRate = mOutRate;
|
|
mOutRate = static_cast<int>(mInRate / aPlaybackRate);
|
|
} else {
|
|
// The playbackRate has been changed before the end of the latency
|
|
// compensation phase. We don't update the mOld* variable. That way, the
|
|
// last playbackRate set is taken into account.
|
|
mBasePosition = GetPosition();
|
|
mBaseOffset = position;
|
|
mPlaybackRateChangeOffset = mWritten;
|
|
mOutRate = static_cast<int>(mInRate / aPlaybackRate);
|
|
}
|
|
}
|
|
|
|
double AudioClock::GetPlaybackRate()
|
|
{
|
|
return static_cast<double>(mInRate) / mOutRate;
|
|
}
|
|
|
|
void AudioClock::SetPreservesPitch(bool aPreservesPitch)
|
|
{
|
|
mPreservesPitch = aPreservesPitch;
|
|
}
|
|
|
|
bool AudioClock::GetPreservesPitch()
|
|
{
|
|
return mPreservesPitch;
|
|
}
|
|
} // namespace mozilla
|