Refactored DecoderBuffer to use unix_hacker_style naming.

media/filters/opus_audio_decoder.cc

 // Copyright (c) 2012 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 "media/filters/opus_audio_decoder.h"
 
 #include "base/bind.h"
 #include "base/callback_helpers.h"
 #include "base/location.h"
 #include "base/message_loop/message_loop_proxy.h"
@@ skipping 18 matching lines @@
   memcpy(&value, data + read_offset, sizeof(value));
   return base::ByteSwapToLE16(value);
 }
 
 // Returns true if the decode result was end of stream.
 static inline bool IsEndOfStream(int decoded_size,
                                  const scoped_refptr<DecoderBuffer>& input) {
   // Two conditions to meet to declare end of stream for this decoder:
   // 1. Opus didn't output anything.
   // 2. An end of stream buffer is received.
-  return decoded_size == 0 && input->IsEndOfStream();
+  return decoded_size == 0 && input->end_of_stream();
 }
 
 // The Opus specification is part of IETF RFC 6716:
 // http://tools.ietf.org/html/rfc6716
 
 // Opus uses Vorbis channel mapping, and Vorbis channel mapping specifies
 // mappings for up to 8 channels. This information is part of the Vorbis I
 // Specification:
 // http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html
 static const int kMaxVorbisChannels = 8;
@@ skipping 33 matching lines @@
   // 8 Channels (7.1)
   //   Front L, Center, Front R, Side L, Side R, Back L, Back R, LFE
   //
   // Channel ordering information is taken from section 4.3.9 of the Vorbis I
   // Specification:
   // http://xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9
 
   // These are the FFmpeg channel layouts expressed using the position of each
   // channel in the output stream from libopus.
   const uint8 kFFmpegChannelLayouts[num_layouts][num_layout_values] = {
-    { 0 },
+    {0},
 
     // Stereo: No reorder.
-    { 0, 1 },
+    {0, 1},
 
     // 3 Channels, from Vorbis order to:
     //  L, R, Center
-    { 0, 2, 1 },
+    {0, 2, 1},
 
     // 4 Channels: No reorder.
-    { 0, 1, 2, 3 },
+    {0, 1, 2, 3},
 
     // 5 Channels, from Vorbis order to:
     //  Front L, Front R, Center, Back L, Back R
-    { 0, 2, 1, 3, 4 },
+    {0, 2, 1, 3, 4},
 
     // 6 Channels (5.1), from Vorbis order to:
     //  Front L, Front R, Center, LFE, Back L, Back R
-    { 0, 2, 1, 5, 3, 4 },
+    {0, 2, 1, 5, 3, 4},
 
     // 7 Channels (6.1), from Vorbis order to:
     //  Front L, Front R, Front Center, LFE, Side L, Side R, Back Center
-    { 0, 2, 1, 6, 3, 4, 5 },
+    {0, 2, 1, 6, 3, 4, 5},
 
     // 8 Channels (7.1), from Vorbis order to:
     //  Front L, Front R, Center, LFE, Back L, Back R, Side L, Side R
-    { 0, 2, 1, 7, 5, 6, 3, 4 },
+    {0, 2, 1, 7, 5, 6, 3, 4},
   };
 
   // Reorder the channels to produce the same ordering as FFmpeg, which is
   // what the pipeline expects.
   const uint8* vorbis_layout_offset = kFFmpegChannelLayouts[num_channels - 1];
   for (int channel = 0; channel < num_channels; ++channel)
     channel_layout[channel] = opus_mapping[vorbis_layout_offset[channel]];
 }
 
 // Opus Header contents:
@@ skipping 18 matching lines @@
 //                - right
 //         - else
 //            - stream = byte-M
 
 // Default audio output channel layout. Used to initialize |stream_map| in
 // OpusHeader, and passed to opus_multistream_decoder_create() when the header
 // does not contain mapping information. The values are valid only for mono and
 // stereo output: Opus streams with more than 2 channels require a stream map.
 static const int kMaxChannelsWithDefaultLayout = 2;
 static const uint8 kDefaultOpusChannelLayout[kMaxChannelsWithDefaultLayout] = {
-    0, 1 };
+  0, 1
+};
 
 // Size of the Opus header excluding optional mapping information.
 static const int kOpusHeaderSize = 19;
 
 // Offset to the channel count byte in the Opus header.
 static const int kOpusHeaderChannelsOffset = 9;
 
 // Offset to the pre-skip value in the Opus header.
 static const int kOpusHeaderSkipSamplesOffset = 10;
 
@@ skipping 10 matching lines @@
 static const int kOpusHeaderNumCoupledOffset = kOpusHeaderNumStreamsOffset + 1;
 static const int kOpusHeaderStreamMapOffset = kOpusHeaderNumStreamsOffset + 2;
 
 struct OpusHeader {
   OpusHeader()
       : channels(0),
         skip_samples(0),
         channel_mapping(0),
         num_streams(0),
         num_coupled(0) {
-    memcpy(stream_map,
-           kDefaultOpusChannelLayout,
-           kMaxChannelsWithDefaultLayout);
+    memcpy(
+        stream_map, kDefaultOpusChannelLayout, kMaxChannelsWithDefaultLayout);
   }
   int channels;
   int skip_samples;
   int channel_mapping;
   int num_streams;
   int num_coupled;
   uint8 stream_map[kMaxVorbisChannels];
 };
 
 // Returns true when able to successfully parse and store Opus header data in
 // data parsed in |header|. Based on opus header parsing code in libopusdec
 // from FFmpeg, and opus_header from Xiph's opus-tools project.
-static void ParseOpusHeader(const uint8* data, int data_size,
+static void ParseOpusHeader(const uint8* data,
+                            int data_size,
                             const AudioDecoderConfig& config,
                             OpusHeader* header) {
   CHECK_GE(data_size, kOpusHeaderSize);
 
   header->channels = *(data + kOpusHeaderChannelsOffset);
 
   CHECK(header->channels > 0 && header->channels <= kMaxVorbisChannels)
       << "invalid channel count in header: " << header->channels;
 
   header->skip_samples =
@@ skipping 29 matching lines @@
     const scoped_refptr<base::MessageLoopProxy>& message_loop)
     : message_loop_(message_loop),
       weak_factory_(this),
       demuxer_stream_(NULL),
       opus_decoder_(NULL),
       bits_per_channel_(0),
       channel_layout_(CHANNEL_LAYOUT_NONE),
       samples_per_second_(0),
       last_input_timestamp_(kNoTimestamp()),
       output_bytes_to_drop_(0),
-      skip_samples_(0) {
-}
+      skip_samples_(0) {}
 
-void OpusAudioDecoder::Initialize(
-    DemuxerStream* stream,
-    const PipelineStatusCB& status_cb,
-    const StatisticsCB& statistics_cb) {
+void OpusAudioDecoder::Initialize(DemuxerStream* stream,
+                                  const PipelineStatusCB& status_cb,
+                                  const StatisticsCB& statistics_cb) {
   DCHECK(message_loop_->BelongsToCurrentThread());
   PipelineStatusCB initialize_cb = BindToCurrentLoop(status_cb);
 
   if (demuxer_stream_) {
     // TODO(scherkus): initialization currently happens more than once in
     // PipelineIntegrationTest.BasicPlayback.
     LOG(ERROR) << "Initialize has already been called.";
     CHECK(false);
   }
 
@@ skipping 46 matching lines @@
   // TODO(scherkus): should we require Stop() to be called? this might end up
   // getting called on a random thread due to refcounting.
   CloseDecoder();
 }
 
 void OpusAudioDecoder::ReadFromDemuxerStream() {
   DCHECK(!read_cb_.is_null());
   demuxer_stream_->Read(base::Bind(&OpusAudioDecoder::BufferReady, weak_this_));
 }
 
-void OpusAudioDecoder::BufferReady(
-    DemuxerStream::Status status,
-    const scoped_refptr<DecoderBuffer>& input) {
+void OpusAudioDecoder::BufferReady(DemuxerStream::Status status,
+                                   const scoped_refptr<DecoderBuffer>& input) {
   DCHECK(message_loop_->BelongsToCurrentThread());
   DCHECK(!read_cb_.is_null());
   DCHECK_EQ(status != DemuxerStream::kOk, !input.get()) << status;
 
   if (status == DemuxerStream::kAborted) {
     DCHECK(!input.get());
     base::ResetAndReturn(&read_cb_).Run(kAborted, NULL);
     return;
   }
 
   if (status == DemuxerStream::kConfigChanged) {
     DCHECK(!input.get());
     DVLOG(1) << "Config changed.";
 
     if (!ConfigureDecoder()) {
       base::ResetAndReturn(&read_cb_).Run(kDecodeError, NULL);
       return;
     }
 
     ResetTimestampState();
     ReadFromDemuxerStream();
     return;
   }
 
   DCHECK_EQ(status, DemuxerStream::kOk);
   DCHECK(input.get());
 
   // Libopus does not buffer output. Decoding is complete when an end of stream
   // input buffer is received.
-  if (input->IsEndOfStream()) {
+  if (input->end_of_stream()) {
     base::ResetAndReturn(&read_cb_).Run(kOk, DataBuffer::CreateEOSBuffer());
     return;
   }
 
   // Make sure we are notified if http://crbug.com/49709 returns.  Issue also
   // occurs with some damaged files.
-  if (input->GetTimestamp() == kNoTimestamp() &&
+  if (input->timestamp() == kNoTimestamp() &&
       output_timestamp_helper_->base_timestamp() == kNoTimestamp()) {
     DVLOG(1) << "Received a buffer without timestamps!";
     base::ResetAndReturn(&read_cb_).Run(kDecodeError, NULL);
     return;
   }
 
   if (last_input_timestamp_ != kNoTimestamp() &&
-      input->GetTimestamp() != kNoTimestamp() &&
-      input->GetTimestamp() < last_input_timestamp_) {
-    base::TimeDelta diff = input->GetTimestamp() - last_input_timestamp_;
+      input->timestamp() != kNoTimestamp() &&
+      input->timestamp() < last_input_timestamp_) {
+    base::TimeDelta diff = input->timestamp() - last_input_timestamp_;
     DVLOG(1) << "Input timestamps are not monotonically increasing! "
-              << " ts " << input->GetTimestamp().InMicroseconds() << " us"
-              << " diff " << diff.InMicroseconds() << " us";
+             << " ts " << input->timestamp().InMicroseconds() << " us"
+             << " diff " << diff.InMicroseconds() << " us";
     base::ResetAndReturn(&read_cb_).Run(kDecodeError, NULL);
     return;
   }
 
-  last_input_timestamp_ = input->GetTimestamp();
+  last_input_timestamp_ = input->timestamp();
 
   scoped_refptr<DataBuffer> output_buffer;
 
   if (!Decode(input, &output_buffer)) {
     base::ResetAndReturn(&read_cb_).Run(kDecodeError, NULL);
     return;
   }
 
   if (output_buffer.get()) {
     // Execute callback to return the decoded audio.
@@ skipping 28 matching lines @@
   if (config.bits_per_channel() != kBitsPerChannel) {
     DLOG(ERROR) << "16 bit samples required.";
     return false;
   }
 
   if (config.is_encrypted()) {
     DLOG(ERROR) << "Encrypted audio stream not supported.";
     return false;
   }
 
-  if (opus_decoder_ &&
-      (bits_per_channel_ != config.bits_per_channel() ||
-       channel_layout_ != config.channel_layout() ||
-       samples_per_second_ != config.samples_per_second())) {
+  if (opus_decoder_ && (bits_per_channel_ != config.bits_per_channel() ||
+                        channel_layout_ != config.channel_layout() ||
+                        samples_per_second_ != config.samples_per_second())) {
     DVLOG(1) << "Unsupported config change :";
-    DVLOG(1) << "\tbits_per_channel : " << bits_per_channel_
-             << " -> " << config.bits_per_channel();
-    DVLOG(1) << "\tchannel_layout : " << channel_layout_
-             << " -> " << config.channel_layout();
-    DVLOG(1) << "\tsample_rate : " << samples_per_second_
-             << " -> " << config.samples_per_second();
+    DVLOG(1) << "\tbits_per_channel : " << bits_per_channel_ << " -> "
+             << config.bits_per_channel();
+    DVLOG(1) << "\tchannel_layout : " << channel_layout_ << " -> "
+             << config.channel_layout();
+    DVLOG(1) << "\tsample_rate : " << samples_per_second_ << " -> "
+             << config.samples_per_second();
     return false;
   }
 
   // Clean up existing decoder if necessary.
   CloseDecoder();
 
   // Allocate the output buffer if necessary.
   if (!output_buffer_)
     output_buffer_.reset(new int16[kMaxOpusOutputPacketSizeSamples]);
 
   // Parse the Opus header.
   OpusHeader opus_header;
-  ParseOpusHeader(config.extra_data(), config.extra_data_size(),
-                  config,
-                  &opus_header);
+  ParseOpusHeader(
+      config.extra_data(), config.extra_data_size(), config, &opus_header);
 
   skip_samples_ = opus_header.skip_samples;
 
   if (skip_samples_ > 0)
     output_bytes_to_drop_ = skip_samples_ * config.bytes_per_frame();
 
   uint8 channel_mapping[kMaxVorbisChannels];
   memcpy(&channel_mapping,
          kDefaultOpusChannelLayout,
          kMaxChannelsWithDefaultLayout);
 
   if (channel_count > kMaxChannelsWithDefaultLayout) {
-    RemapOpusChannelLayout(opus_header.stream_map,
-                           channel_count,
-                           channel_mapping);
+    RemapOpusChannelLayout(
+        opus_header.stream_map, channel_count, channel_mapping);
   }
 
   // Init Opus.
   int status = OPUS_INVALID_STATE;
   opus_decoder_ = opus_multistream_decoder_create(config.samples_per_second(),
                                                   channel_count,
                                                   opus_header.num_streams,
                                                   opus_header.num_coupled,
                                                   channel_mapping,
                                                   &status);
@@ skipping 24 matching lines @@
 void OpusAudioDecoder::ResetTimestampState() {
   output_timestamp_helper_->SetBaseTimestamp(kNoTimestamp());
   last_input_timestamp_ = kNoTimestamp();
   output_bytes_to_drop_ = 0;
 }
 
 bool OpusAudioDecoder::Decode(const scoped_refptr<DecoderBuffer>& input,
                               scoped_refptr<DataBuffer>* output_buffer) {
   const int samples_decoded =
       opus_multistream_decode(opus_decoder_,
-                              input->GetData(), input->GetDataSize(),
+                              input->data(),
+                              input->data_size(),
                               &output_buffer_[0],
                               kMaxOpusOutputPacketSizeSamples,
                               0);
   if (samples_decoded < 0) {
     LOG(ERROR) << "opus_multistream_decode failed for"
-               << " timestamp: " << input->GetTimestamp().InMicroseconds()
-               << " us, duration: " << input->GetDuration().InMicroseconds()
-               << " us, packet size: " << input->GetDataSize() << " bytes with"
+               << " timestamp: " << input->timestamp().InMicroseconds()
+               << " us, duration: " << input->duration().InMicroseconds()
+               << " us, packet size: " << input->data_size() << " bytes with"
                << " status: " << opus_strerror(samples_decoded);
     return false;
   }
 
   uint8* decoded_audio_data = reinterpret_cast<uint8*>(&output_buffer_[0]);
-  int decoded_audio_size = samples_decoded *
+  int decoded_audio_size =
+      samples_decoded *
       demuxer_stream_->audio_decoder_config().bytes_per_frame();
   DCHECK_LE(decoded_audio_size, kMaxOpusOutputPacketSizeBytes);
 
   if (output_timestamp_helper_->base_timestamp() == kNoTimestamp() &&
-      !input->IsEndOfStream()) {
-    DCHECK(input->GetTimestamp() != kNoTimestamp());
-    output_timestamp_helper_->SetBaseTimestamp(input->GetTimestamp());
+      !input->end_of_stream()) {
+    DCHECK(input->timestamp() != kNoTimestamp());
+    output_timestamp_helper_->SetBaseTimestamp(input->timestamp());
   }
 
   if (decoded_audio_size > 0 && output_bytes_to_drop_ > 0) {
     int dropped_size = std::min(decoded_audio_size, output_bytes_to_drop_);
     DCHECK_EQ(dropped_size % kBytesPerChannel, 0);
     decoded_audio_data += dropped_size;
     decoded_audio_size -= dropped_size;
     output_bytes_to_drop_ -= dropped_size;
   }
 
   if (decoded_audio_size > 0) {
     // Copy the audio samples into an output buffer.
-    *output_buffer = DataBuffer::CopyFrom(
-        decoded_audio_data, decoded_audio_size);
+    *output_buffer =
+        DataBuffer::CopyFrom(decoded_audio_data, decoded_audio_size);
     (*output_buffer)->set_timestamp(output_timestamp_helper_->GetTimestamp());
     (*output_buffer)->set_duration(
         output_timestamp_helper_->GetDuration(decoded_audio_size));
     output_timestamp_helper_->AddBytes(decoded_audio_size);
   }
 
   // Decoding finished successfully, update statistics.
   PipelineStatistics statistics;
   statistics.audio_bytes_decoded = decoded_audio_size;
   statistics_cb_.Run(statistics);
 
   return true;
 }
 
 }  // namespace media