Chromium Code Reviews| Index: services/media/framework_ffmpeg/ffmpeg_audio_decoder.cc |
| diff --git a/services/media/framework_ffmpeg/ffmpeg_audio_decoder.cc b/services/media/framework_ffmpeg/ffmpeg_audio_decoder.cc |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..a2d425e6c3d7298494355b04a8edb0d370c53257 |
| --- /dev/null |
| +++ b/services/media/framework_ffmpeg/ffmpeg_audio_decoder.cc |
| @@ -0,0 +1,205 @@ |
| +// Copyright 2016 The Chromium Authors. All rights reserved. |
| +// Use of this source code is governed by a BSD-style license that can be |
| +// found in the LICENSE file. |
| + |
| +#include "base/logging.h" |
| +#include "services/media/framework_ffmpeg/ffmpeg_audio_decoder.h" |
| + |
| +namespace mojo { |
| +namespace media { |
| + |
| +FfmpegAudioDecoder::FfmpegAudioDecoder(AvCodecContextPtr av_codec_context) : |
| + FfmpegDecoderBase(std::move(av_codec_context)) { |
| + DCHECK(context()); |
| + DCHECK(context()->channels > 0); |
| + |
| + context()->opaque = this; |
| + context()->get_buffer2 = AllocateBufferForAvFrame; |
| + context()->refcounted_frames = 1; |
| + |
| + if (av_sample_fmt_is_planar(context()->sample_fmt)) { |
| + // Prepare for interleaving. |
| + stream_type_ = output_stream_type(); |
| + lpcm_util_ = LpcmUtil::Create(*stream_type_->lpcm()); |
| + } |
| +} |
| + |
| +FfmpegAudioDecoder::~FfmpegAudioDecoder() {} |
| + |
| +int FfmpegAudioDecoder::Decode( |
| + PayloadAllocator* allocator, |
| + bool* frame_decoded_out) { |
| + DCHECK(allocator); |
| + DCHECK(frame_decoded_out); |
| + DCHECK(context()); |
| + DCHECK(frame()); |
| + |
| + // Use the provided allocator (for allocations in AllocateBufferForAvFrame) |
| + // unless we intend to interleave later, in which case use the default |
| + // allocator. We'll interleave into a buffer from the provided allocator |
| + // in CreateOutputPacket. |
| + allocator_ = lpcm_util_ ? PayloadAllocator::GetDefault() : allocator; |
| + |
| + int frame_decoded = 0; |
| + int input_bytes_used = avcodec_decode_audio4( |
| + context().get(), |
| + frame().get(), |
| + &frame_decoded, |
| + &packet()); |
| + *frame_decoded_out = frame_decoded != 0; |
| + |
| + // We're done with this allocator. |
| + allocator_ = nullptr; |
| + |
| + return input_bytes_used; |
| +} |
| + |
| +PacketPtr FfmpegAudioDecoder::CreateOutputPacket(PayloadAllocator* allocator) { |
| + DCHECK(allocator); |
| + DCHECK(frame()); |
| + |
| + int64_t presentation_time = frame()->pts; |
| + if (presentation_time == AV_NOPTS_VALUE) { |
| + // TODO(dalesat): Adjust next_presentation_time_ for seek/non-zero start. |
| + presentation_time = next_presentation_time_; |
| + next_presentation_time_ += frame()->nb_samples; |
| + } |
| + |
| + uint64_t payload_size; |
| + void *payload_buffer; |
| + |
| + AvBufferContext* av_buffer_context = |
| + reinterpret_cast<AvBufferContext*>(av_buffer_get_opaque(frame()->buf[0])); |
| + |
| + if (lpcm_util_) { |
| + // We need to interleave. The non-interleaved frames are in a buffer that |
| + // was allocated from the default allocator. That buffer will get released |
| + // later in ReleaseBufferForAvFrame. We need a new buffer for the |
| + // interleaved frames, which we get from the provided allocator. |
| + DCHECK(stream_type_); |
| + DCHECK(stream_type_->lpcm()); |
| + payload_size = stream_type_->lpcm()->min_buffer_size(frame()->nb_samples); |
| + payload_buffer = allocator->AllocatePayloadBuffer(payload_size); |
| + |
| + lpcm_util_->Interleave( |
| + av_buffer_context->buffer(), |
| + av_buffer_context->size(), |
| + payload_buffer, |
| + frame()->nb_samples); |
| + } else { |
| + // We don't need to interleave. The interleaved frames are in a buffer that |
| + // was allocated from the correct allocator. We take ownership of the buffer |
| + // by calling Release here so that ReleaseBufferForAvFrame won't release it. |
| + payload_size = av_buffer_context->size(); |
| + payload_buffer = av_buffer_context->Release(); |
| + } |
| + |
| + return Packet::Create( |
| + presentation_time, |
| + frame()->nb_samples, |
| + false, // The base class is responsible for end-of-stream. |
| + payload_size, |
| + payload_buffer, |
| + allocator); |
| +} |
| + |
| +PacketPtr FfmpegAudioDecoder::CreateOutputEndOfStreamPacket() { |
| + return Packet::CreateEndOfStream(next_presentation_time_); |
| +} |
| + |
| +int FfmpegAudioDecoder::AllocateBufferForAvFrame( |
| + AVCodecContext* av_codec_context, |
| + AVFrame* av_frame, |
| + int flags) { |
| + // CODEC_CAP_DR1 is required in order to do allocation this way. |
| + DCHECK(av_codec_context->codec->capabilities & CODEC_CAP_DR1); |
| + |
| + FfmpegAudioDecoder* self = |
| + reinterpret_cast<FfmpegAudioDecoder*>(av_codec_context->opaque); |
| + DCHECK(self); |
| + DCHECK(self->allocator_); |
| + |
| + AVSampleFormat av_sample_format = |
| + static_cast<AVSampleFormat>(av_frame->format); |
| + |
| + int buffer_size = av_samples_get_buffer_size( |
| + &av_frame->linesize[0], |
| + av_codec_context->channels, |
| + av_frame->nb_samples, |
| + av_sample_format, |
| + FfmpegAudioDecoder::kChannelAlign); |
| + if (buffer_size < 0) { |
| + LOG(WARNING) << "av_samples_get_buffer_size failed"; |
| + return buffer_size; |
| + } |
| + |
| + AvBufferContext* av_buffer_context = |
| + new AvBufferContext(buffer_size, self->allocator_); |
| + uint8_t* buffer = av_buffer_context->buffer(); |
| + |
| + if (!av_sample_fmt_is_planar(av_sample_format)) { |
| + // Samples are interleaved. There's just one buffer. |
| + av_frame->data[0] = buffer; |
| + } else { |
| + // Samples are not interleaved. There's one buffer per channel. |
| + int channels = av_codec_context->channels; |
| + int bytes_per_channel = buffer_size / channels; |
| + uint8_t* channel_buffer = buffer; |
| + |
| + DCHECK(buffer != nullptr || bytes_per_channel == 0); |
| + |
| + if (channels <= AV_NUM_DATA_POINTERS) { |
| + // The buffer pointers will fit in av_frame->data. |
| + DCHECK_EQ(av_frame->extended_data, av_frame->data); |
| + for (int channel = 0; channel < channels; ++channel) { |
| + av_frame->data[channel] = channel_buffer; |
| + channel_buffer += bytes_per_channel; |
| + } |
| + } else { |
| + // Too many channels for av_frame->data. We have to use |
| + // av_frame->extended_data |
| + av_frame->extended_data = static_cast<uint8_t**>( |
| + av_malloc(channels * sizeof(*av_frame->extended_data))); |
| + |
| + // The first AV_NUM_DATA_POINTERS go in both data and extended_data. |
| + int channel = 0; |
| + for (; channel < AV_NUM_DATA_POINTERS; ++channel) { |
| + av_frame->extended_data[channel] = av_frame->data[channel] = |
| + channel_buffer; |
| + channel_buffer += bytes_per_channel; |
| + } |
| + |
| + // The rest go only in extended_data. |
| + for (; channel < channels; ++channel) { |
| + av_frame->extended_data[channel] = channel_buffer; |
| + channel_buffer += bytes_per_channel; |
| + } |
| + } |
| + } |
| + |
| + av_frame->buf[0] = av_buffer_create( |
| + buffer, |
| + buffer_size, |
| + ReleaseBufferForAvFrame, |
| + av_buffer_context, |
| + 0); // flags |
| + |
| + return 0; |
| +} |
| + |
| +void FfmpegAudioDecoder::ReleaseBufferForAvFrame( |
| + void* opaque, |
| + uint8_t* buffer) { |
| + AvBufferContext* av_buffer_context = |
| + reinterpret_cast<AvBufferContext*>(opaque); |
| + DCHECK(av_buffer_context); |
| + // Either this buffer has already been released to someone else's ownership, |
| + // or it's the same as the buffer parameter. |
|
johngro
2016/03/03 23:03:16
this still makes me vaguely uneasy... The FFmpeg
|
| + DCHECK( |
| + av_buffer_context->buffer() == nullptr || |
| + av_buffer_context->buffer() == buffer); |
| + delete av_buffer_context; |
| +} |
| + |
| +} // namespace media |
| +} // namespace mojo |