/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */ /* vim:set ts=2 sw=2 sts=2 et cindent: */ /* This Source Code Form is subject to the terms of the Mozilla Public * License, v. 2.0. If a copy of the MPL was not distributed with this * file, You can obtain one at http://mozilla.org/MPL/2.0/. */ #include "MediaTaskQueue.h" #include "FFmpegRuntimeLinker.h" #include "FFmpegAudioDecoder.h" #define MAX_CHANNELS 16 namespace mozilla { FFmpegAudioDecoder::FFmpegAudioDecoder( FlushableMediaTaskQueue* aTaskQueue, MediaDataDecoderCallback* aCallback, const AudioInfo& aConfig) : FFmpegDataDecoder(aTaskQueue, GetCodecId(aConfig.mMimeType)) , mCallback(aCallback) { MOZ_COUNT_CTOR(FFmpegAudioDecoder); // Use a new MediaByteBuffer as the object will be modified during initialization. mExtraData = new MediaByteBuffer; mExtraData->AppendElements(*aConfig.mCodecSpecificConfig); } nsresult FFmpegAudioDecoder::Init() { nsresult rv = FFmpegDataDecoder::Init(); NS_ENSURE_SUCCESS(rv, rv); return NS_OK; } void FFmpegAudioDecoder::InitCodecContext() { MOZ_ASSERT(mCodecContext); // We do not want to set this value to 0 as FFmpeg by default will // use the number of cores, which with our mozlibavutil get_cpu_count // isn't implemented. mCodecContext->thread_count = 1; // FFmpeg takes this as a suggestion for what format to use for audio samples. uint32_t major, minor, micro; FFmpegRuntimeLinker::GetVersion(major, minor, micro); // LibAV 0.8 produces rubbish float interleaved samples, request 16 bits audio. mCodecContext->request_sample_fmt = (major == 53) ? AV_SAMPLE_FMT_S16 : AV_SAMPLE_FMT_FLT; } static AudioDataValue* CopyAndPackAudio(AVFrame* aFrame, uint32_t aNumChannels, uint32_t aNumAFrames) { MOZ_ASSERT(aNumChannels <= MAX_CHANNELS); nsAutoArrayPtr audio( new AudioDataValue[aNumChannels * aNumAFrames]); if (aFrame->format == AV_SAMPLE_FMT_FLT) { // Audio data already packed. No need to do anything other than copy it // into a buffer we own. memcpy(audio, aFrame->data[0], aNumChannels * aNumAFrames * sizeof(AudioDataValue)); } else if (aFrame->format == AV_SAMPLE_FMT_FLTP) { // Planar audio data. Pack it into something we can understand. AudioDataValue* tmp = audio; AudioDataValue** data = reinterpret_cast(aFrame->data); for (uint32_t frame = 0; frame < aNumAFrames; frame++) { for (uint32_t channel = 0; channel < aNumChannels; channel++) { *tmp++ = data[channel][frame]; } } } else if (aFrame->format == AV_SAMPLE_FMT_S16) { // Audio data already packed. Need to convert from S16 to 32 bits Float AudioDataValue* tmp = audio; int16_t* data = reinterpret_cast(aFrame->data)[0]; for (uint32_t frame = 0; frame < aNumAFrames; frame++) { for (uint32_t channel = 0; channel < aNumChannels; channel++) { *tmp++ = AudioSampleToFloat(*data++); } } } else if (aFrame->format == AV_SAMPLE_FMT_S16P) { // Planar audio data. Convert it from S16 to 32 bits float // and pack it into something we can understand. AudioDataValue* tmp = audio; int16_t** data = reinterpret_cast(aFrame->data); for (uint32_t frame = 0; frame < aNumAFrames; frame++) { for (uint32_t channel = 0; channel < aNumChannels; channel++) { *tmp++ = AudioSampleToFloat(data[channel][frame]); } } } return audio.forget(); } void FFmpegAudioDecoder::DecodePacket(MediaRawData* aSample) { AVPacket packet; av_init_packet(&packet); packet.data = const_cast(aSample->Data()); packet.size = aSample->Size(); if (!PrepareFrame()) { NS_WARNING("FFmpeg audio decoder failed to allocate frame."); mCallback->Error(); return; } int64_t samplePosition = aSample->mOffset; Microseconds pts = aSample->mTime; while (packet.size > 0) { int decoded; int bytesConsumed = avcodec_decode_audio4(mCodecContext, mFrame, &decoded, &packet); if (bytesConsumed < 0) { NS_WARNING("FFmpeg audio decoder error."); mCallback->Error(); return; } if (decoded) { uint32_t numChannels = mCodecContext->channels; uint32_t samplingRate = mCodecContext->sample_rate; nsAutoArrayPtr audio( CopyAndPackAudio(mFrame, numChannels, mFrame->nb_samples)); CheckedInt duration = FramesToUsecs(mFrame->nb_samples, samplingRate); if (!duration.isValid()) { NS_WARNING("Invalid count of accumulated audio samples"); mCallback->Error(); return; } nsRefPtr data = new AudioData(samplePosition, pts, duration.value(), mFrame->nb_samples, audio.forget(), numChannels, samplingRate); mCallback->Output(data); pts += duration.value(); } packet.data += bytesConsumed; packet.size -= bytesConsumed; samplePosition += bytesConsumed; } if (mTaskQueue->IsEmpty()) { mCallback->InputExhausted(); } } nsresult FFmpegAudioDecoder::Input(MediaRawData* aSample) { mTaskQueue->Dispatch(NS_NewRunnableMethodWithArg >( this, &FFmpegAudioDecoder::DecodePacket, nsRefPtr(aSample))); return NS_OK; } nsresult FFmpegAudioDecoder::Drain() { mTaskQueue->AwaitIdle(); mCallback->DrainComplete(); return Flush(); } AVCodecID FFmpegAudioDecoder::GetCodecId(const nsACString& aMimeType) { if (aMimeType.EqualsLiteral("audio/mpeg")) { return AV_CODEC_ID_MP3; } if (aMimeType.EqualsLiteral("audio/mp4a-latm")) { return AV_CODEC_ID_AAC; } return AV_CODEC_ID_NONE; } FFmpegAudioDecoder::~FFmpegAudioDecoder() { MOZ_COUNT_DTOR(FFmpegAudioDecoder); } } // namespace mozilla