mirror of
https://gitlab.winehq.org/wine/wine-gecko.git
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118 lines
3.7 KiB
C++
118 lines
3.7 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
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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 file,
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* You can obtain one at http://mozilla.org/MPL/2.0/. */
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#include "TrackEncoder.h"
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#include "MediaStreamGraph.h"
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#include "AudioChannelFormat.h"
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#undef LOG
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#ifdef MOZ_WIDGET_GONK
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#include <android/log.h>
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#define LOG(args...) __android_log_print(ANDROID_LOG_INFO, "MediakEncoder", ## args);
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#else
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#define LOG(args, ...)
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#endif
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namespace mozilla {
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#define MAX_FRAMES_TO_DROP 48000
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void
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AudioTrackEncoder::NotifyQueuedTrackChanges(MediaStreamGraph* aGraph,
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TrackID aID,
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TrackRate aTrackRate,
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TrackTicks aTrackOffset,
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uint32_t aTrackEvents,
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const MediaSegment& aQueuedMedia)
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{
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AudioSegment* audio = const_cast<AudioSegment*>
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(static_cast<const AudioSegment*>(&aQueuedMedia));
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// Check and initialize parameters for codec encoder.
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if (!mInitialized) {
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AudioSegment::ChunkIterator iter(*audio);
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while (!iter.IsEnded()) {
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AudioChunk chunk = *iter;
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if (chunk.mBuffer) {
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Init(chunk.mChannelData.Length(), aTrackRate);
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break;
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}
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iter.Next();
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}
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}
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// Append and consume this raw segment.
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AppendAudioSegment(audio);
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// The stream has stopped and reached the end of track.
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if (aTrackEvents == MediaStreamListener::TRACK_EVENT_ENDED) {
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LOG("[AudioTrackEncoder]: Receive TRACK_EVENT_ENDED .");
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NotifyEndOfStream();
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}
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}
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void
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AudioTrackEncoder::NotifyRemoved(MediaStreamGraph* aGraph)
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{
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// In case that MediaEncoder does not receive a TRACK_EVENT_ENDED event.
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LOG("[AudioTrackEncoder]: NotifyRemoved.");
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NotifyEndOfStream();
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}
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nsresult
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AudioTrackEncoder::AppendAudioSegment(MediaSegment* aSegment)
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{
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// Drop the in-coming segment if buffer(mRawSegment) is overflow.
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ReentrantMonitorAutoEnter mon(mReentrantMonitor);
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AudioSegment* audio = static_cast<AudioSegment*>(aSegment);
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AudioSegment::ChunkIterator iter(*audio);
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if (mRawSegment->GetDuration() < MAX_FRAMES_TO_DROP) {
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while(!iter.IsEnded()) {
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AudioChunk chunk = *iter;
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if (chunk.mBuffer) {
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mRawSegment->AppendAndConsumeChunk(&chunk);
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}
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iter.Next();
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}
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if (mRawSegment->GetDuration() >= GetPacketDuration()) {
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mReentrantMonitor.NotifyAll();
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}
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}
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#ifdef DEBUG
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else {
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LOG("[AudioTrackEncoder]: A segment has dropped!");
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}
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#endif
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return NS_OK;
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}
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static const int AUDIO_PROCESSING_FRAMES = 640; /* > 10ms of 48KHz audio */
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static const uint8_t gZeroChannel[MAX_AUDIO_SAMPLE_SIZE*AUDIO_PROCESSING_FRAMES] = {0};
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void
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AudioTrackEncoder::InterleaveTrackData(AudioChunk& aChunk,
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int32_t aDuration,
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uint32_t aOutputChannels,
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AudioDataValue* aOutput)
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{
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if (aChunk.mChannelData.Length() < aOutputChannels) {
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// Up-mix. This might make the mChannelData have more than aChannels.
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AudioChannelsUpMix(&aChunk.mChannelData, aOutputChannels, gZeroChannel);
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}
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if (aChunk.mChannelData.Length() > aOutputChannels) {
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DownmixAndInterleave(aChunk.mChannelData, aChunk.mBufferFormat, aDuration,
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aChunk.mVolume, mChannels, aOutput);
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} else {
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InterleaveAndConvertBuffer(aChunk.mChannelData.Elements(),
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aChunk.mBufferFormat, aDuration, aChunk.mVolume,
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mChannels, aOutput);
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}
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}
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}
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