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Unified Diff: media/cdm/ppapi/ffmpeg_cdm_audio_decoder.cc

Issue 26956002: Plumb support for audio sample formats. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Fix cast. Created 7 years, 2 months ago
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Index: media/cdm/ppapi/ffmpeg_cdm_audio_decoder.cc
diff --git a/media/cdm/ppapi/ffmpeg_cdm_audio_decoder.cc b/media/cdm/ppapi/ffmpeg_cdm_audio_decoder.cc
index 3e2ca6504bb9adedb07f28c1322ee3f6b190c23e..97d04e867cf9bc09f6676a1ce2c1ceead7258a2f 100644
--- a/media/cdm/ppapi/ffmpeg_cdm_audio_decoder.cc
+++ b/media/cdm/ppapi/ffmpeg_cdm_audio_decoder.cc
@@ -80,10 +80,59 @@ static void CdmAudioDecoderConfigToAVCodecContext(
}
}
+static cdm::AudioFormat AVSampleFormatToCdmAudioFormat(
+ AVSampleFormat sample_format) {
+ switch (sample_format) {
+ case AV_SAMPLE_FMT_U8:
+ return cdm::kAudioFormatU8;
+ case AV_SAMPLE_FMT_S16:
+ return cdm::kAudioFormatS16;
+ case AV_SAMPLE_FMT_S32:
+ return cdm::kAudioFormatS32;
+ case AV_SAMPLE_FMT_FLT:
+ return cdm::kAudioFormatF32;
+ case AV_SAMPLE_FMT_S16P:
+ return cdm::kAudioFormatPlanarS16;
+ case AV_SAMPLE_FMT_FLTP:
+ return cdm::kAudioFormatPlanarF32;
+ default:
+ DVLOG(1) << "Unknown AVSampleFormat: " << sample_format;
+ }
+ return cdm::kUnknownAudioFormat;
+}
+
+static void CopySamples(cdm::AudioFormat cdm_format,
+ int decoded_audio_size,
+ const AVFrame& av_frame,
+ uint8_t* output_buffer) {
+ switch (cdm_format) {
+ case cdm::kAudioFormatU8:
+ case cdm::kAudioFormatS16:
+ case cdm::kAudioFormatS32:
+ case cdm::kAudioFormatF32:
+ memcpy(output_buffer, av_frame.data[0], decoded_audio_size);
+ break;
+ case cdm::kAudioFormatPlanarS16:
+ case cdm::kAudioFormatPlanarF32: {
+ const int decoded_size_per_channel =
+ decoded_audio_size / av_frame.channels;
+ for (int i = 0; i < av_frame.channels; ++i) {
+ memcpy(output_buffer,
+ av_frame.extended_data[i],
+ decoded_size_per_channel);
+ output_buffer += decoded_size_per_channel;
+ }
+ break;
+ }
+ default:
+ NOTREACHED() << "Unsupported CDM Audio Format!";
+ memset(output_buffer, 0, decoded_audio_size);
+ }
+}
+
FFmpegCdmAudioDecoder::FFmpegCdmAudioDecoder(cdm::Host* host)
: is_initialized_(false),
host_(host),
- bits_per_channel_(0),
samples_per_second_(0),
channels_(0),
av_sample_format_(0),
@@ -98,7 +147,6 @@ FFmpegCdmAudioDecoder::~FFmpegCdmAudioDecoder() {
bool FFmpegCdmAudioDecoder::Initialize(const cdm::AudioDecoderConfig& config) {
DVLOG(1) << "Initialize()";
-
if (!IsValidConfig(config)) {
LOG(ERROR) << "Initialize(): invalid audio decoder configuration.";
return false;
@@ -131,27 +179,12 @@ bool FFmpegCdmAudioDecoder::Initialize(const cdm::AudioDecoderConfig& config) {
return false;
}
- // Some codecs will only output float data, so we need to convert to integer
- // before returning the decoded buffer.
- if (codec_context_->sample_fmt == AV_SAMPLE_FMT_FLTP ||
- codec_context_->sample_fmt == AV_SAMPLE_FMT_FLT) {
- // Preallocate the AudioBus for float conversions. We can treat interleaved
- // float data as a single planar channel since our output is expected in an
- // interleaved format anyways.
- int channels = codec_context_->channels;
- if (codec_context_->sample_fmt == AV_SAMPLE_FMT_FLT)
- channels = 1;
- converter_bus_ = AudioBus::CreateWrapper(channels);
- }
-
// Success!
av_frame_.reset(avcodec_alloc_frame());
- bits_per_channel_ = config.bits_per_channel;
samples_per_second_ = config.samples_per_second;
- bytes_per_frame_ = codec_context_->channels * bits_per_channel_ / 8;
+ bytes_per_frame_ = codec_context_->channels * config.bits_per_channel / 8;
output_timestamp_helper_.reset(
new AudioTimestampHelper(config.samples_per_second));
- serialized_audio_frames_.reserve(bytes_per_frame_ * samples_per_second_);
is_initialized_ = true;
// Store initial values to guard against midstream configuration changes.
@@ -190,7 +223,7 @@ cdm::Status FFmpegCdmAudioDecoder::DecodeBuffer(
const uint8_t* compressed_buffer,
int32_t compressed_buffer_size,
int64_t input_timestamp,
- cdm::AudioFrames_1* decoded_frames) {
+ cdm::AudioFrames* decoded_frames) {
DVLOG(1) << "DecodeBuffer()";
const bool is_end_of_stream = !compressed_buffer;
base::TimeDelta timestamp =
@@ -226,6 +259,12 @@ cdm::Status FFmpegCdmAudioDecoder::DecodeBuffer(
packet.data = const_cast<uint8_t*>(compressed_buffer);
packet.size = compressed_buffer_size;
+ // Tell the CDM what AudioFormat we're using.
+ const cdm::AudioFormat cdm_format = AVSampleFormatToCdmAudioFormat(
+ static_cast<AVSampleFormat>(av_sample_format_));
+ DCHECK_NE(cdm_format, cdm::kUnknownAudioFormat);
+ decoded_frames->SetFormat(cdm_format);
+
// Each audio packet may contain several frames, so we must call the decoder
// until we've exhausted the packet. Regardless of the packet size we always
// want to hand it to the decoder at least once, otherwise we would end up
@@ -289,76 +328,63 @@ cdm::Status FFmpegCdmAudioDecoder::DecodeBuffer(
decoded_audio_size = av_samples_get_buffer_size(
NULL, codec_context_->channels, av_frame_->nb_samples,
codec_context_->sample_fmt, 1);
- // If we're decoding into float, adjust audio size.
- if (converter_bus_ && bits_per_channel_ / 8 != sizeof(float)) {
- DCHECK(codec_context_->sample_fmt == AV_SAMPLE_FMT_FLT ||
- codec_context_->sample_fmt == AV_SAMPLE_FMT_FLTP);
- decoded_audio_size *=
- static_cast<float>(bits_per_channel_ / 8) / sizeof(float);
- }
}
- int start_sample = 0;
if (decoded_audio_size > 0 && output_bytes_to_drop_ > 0) {
DCHECK_EQ(decoded_audio_size % bytes_per_frame_, 0)
<< "Decoder didn't output full frames";
int dropped_size = std::min(decoded_audio_size, output_bytes_to_drop_);
- start_sample = dropped_size / bytes_per_frame_;
decoded_audio_size -= dropped_size;
output_bytes_to_drop_ -= dropped_size;
}
- scoped_refptr<DataBuffer> output;
if (decoded_audio_size > 0) {
DCHECK_EQ(decoded_audio_size % bytes_per_frame_, 0)
<< "Decoder didn't output full frames";
- // Convert float data using an AudioBus.
- if (converter_bus_) {
- // Setup the AudioBus as a wrapper of the AVFrame data and then use
- // AudioBus::ToInterleaved() to convert the data as necessary.
- int skip_frames = start_sample;
- int total_frames = av_frame_->nb_samples;
- int frames_to_interleave = decoded_audio_size / bytes_per_frame_;
- if (codec_context_->sample_fmt == AV_SAMPLE_FMT_FLT) {
- DCHECK_EQ(converter_bus_->channels(), 1);
- total_frames *= codec_context_->channels;
- skip_frames *= codec_context_->channels;
- frames_to_interleave *= codec_context_->channels;
- }
+ base::TimeDelta output_timestamp =
+ output_timestamp_helper_->GetTimestamp();
+ output_timestamp_helper_->AddFrames(decoded_audio_size /
+ bytes_per_frame_);
- converter_bus_->set_frames(total_frames);
- for (int i = 0; i < converter_bus_->channels(); ++i) {
- converter_bus_->SetChannelData(i, reinterpret_cast<float*>(
- av_frame_->extended_data[i]));
+ // If we've exhausted the packet in the first decode we can write directly
+ // into the frame buffer instead of a multistep serialization approach.
+ if (serialized_audio_frames_.empty() && !packet.size) {
+ const uint32_t buffer_size = decoded_audio_size + sizeof(int64) * 2;
+ decoded_frames->SetFrameBuffer(host_->Allocate(buffer_size));
+ if (!decoded_frames->FrameBuffer()) {
+ LOG(ERROR) << "DecodeBuffer() cdm::Host::Allocate failed.";
+ return cdm::kDecodeError;
}
+ decoded_frames->FrameBuffer()->SetSize(buffer_size);
+ uint8_t* output_buffer = decoded_frames->FrameBuffer()->Data();
- output = new DataBuffer(decoded_audio_size);
- output->set_data_size(decoded_audio_size);
+ const int64 timestamp = output_timestamp.InMicroseconds();
+ memcpy(output_buffer, &timestamp, sizeof(timestamp));
+ output_buffer += sizeof(timestamp);
- DCHECK_EQ(frames_to_interleave, converter_bus_->frames() - skip_frames);
- converter_bus_->ToInterleavedPartial(
- skip_frames, frames_to_interleave, bits_per_channel_ / 8,
- output->writable_data());
- } else {
- output = DataBuffer::CopyFrom(
- av_frame_->extended_data[0] + start_sample * bytes_per_frame_,
- decoded_audio_size);
- }
+ const int64 output_size = decoded_audio_size;
+ memcpy(output_buffer, &output_size, sizeof(output_size));
+ output_buffer += sizeof(output_size);
- base::TimeDelta output_timestamp =
- output_timestamp_helper_->GetTimestamp();
- output_timestamp_helper_->AddFrames(decoded_audio_size /
- bytes_per_frame_);
+ // Copy the samples and return success.
+ CopySamples(
+ cdm_format, decoded_audio_size, *av_frame_, output_buffer);
+ return cdm::kSuccess;
+ }
- // Serialize the audio samples into |serialized_audio_frames_|.
+ // There are still more frames to decode, so we need to serialize them in
+ // a secondary buffer since we don't know their sizes ahead of time (which
+ // is required to allocate the FrameBuffer object).
SerializeInt64(output_timestamp.InMicroseconds());
- SerializeInt64(output->data_size());
- serialized_audio_frames_.insert(
- serialized_audio_frames_.end(),
- output->data(),
- output->data() + output->data_size());
+ SerializeInt64(decoded_audio_size);
+
+ const size_t previous_size = serialized_audio_frames_.size();
+ serialized_audio_frames_.resize(previous_size + decoded_audio_size);
+ uint8_t* output_buffer = &serialized_audio_frames_[0] + previous_size;
+ CopySamples(
+ cdm_format, decoded_audio_size, *av_frame_, output_buffer);
}
} while (packet.size > 0);
@@ -395,7 +421,7 @@ void FFmpegCdmAudioDecoder::ReleaseFFmpegResources() {
}
void FFmpegCdmAudioDecoder::SerializeInt64(int64 value) {
- int previous_size = serialized_audio_frames_.size();
+ const size_t previous_size = serialized_audio_frames_.size();
serialized_audio_frames_.resize(previous_size + sizeof(value));
memcpy(&serialized_audio_frames_[0] + previous_size, &value, sizeof(value));
}
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