Cleanup OPUS decoder. Remove extraneous transforms and copies.

media/filters/opus_audio_decoder.cc

Index: media/filters/opus_audio_decoder.cc
diff --git a/media/filters/opus_audio_decoder.cc b/media/filters/opus_audio_decoder.cc
index 37e1abda698328542350b510558a1763cda60392..6adae4b92f3ab99e5178c3702cd6ce207ebfaaf0 100644
--- a/media/filters/opus_audio_decoder.cc
+++ b/media/filters/opus_audio_decoder.cc
@@ -25,7 +25,6 @@
 namespace media {
 
 static uint16 ReadLE16(const uint8* data, size_t data_size, int read_offset) {
-  DCHECK(data);
   uint16 value = 0;
   DCHECK_LE(read_offset + sizeof(value), data_size);
   memcpy(&value, data + read_offset, sizeof(value));
@@ -46,15 +45,8 @@ static int TimeDeltaToAudioFrames(base::TimeDelta time_delta,
 // http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html
 static const int kMaxVorbisChannels = 8;
 
-// Opus allows for decode of S16 or float samples. OpusAudioDecoder always uses
-// S16 samples.
-static const int kBitsPerChannel = 16;
-static const int kBytesPerChannel = kBitsPerChannel / 8;
-
 // Maximum packet size used in Xiph's opusdec and FFmpeg's libopusdec.
-static const int kMaxOpusOutputPacketSizeSamples = 960 * 6 * kMaxVorbisChannels;
-static const int kMaxOpusOutputPacketSizeBytes =
-    kMaxOpusOutputPacketSizeSamples * kBytesPerChannel;
+static const int kMaxOpusOutputPacketSizeSamples = 960 * 6;
 
 static void RemapOpusChannelLayout(const uint8* opus_mapping,
                                    int num_channels,
@@ -206,13 +198,16 @@ struct OpusExtraData {
 static bool ParseOpusExtraData(const uint8* data, int data_size,
                                const AudioDecoderConfig& config,
                                OpusExtraData* extra_data) {
-  if (data_size < kOpusExtraDataSize)
+  if (data_size < kOpusExtraDataSize) {
+    DLOG(ERROR) << "Extra data size is too small:" << data_size;
     return false;
+  }
 
   extra_data->channels = *(data + kOpusExtraDataChannelsOffset);
 
   if (extra_data->channels <= 0 || extra_data->channels > kMaxVorbisChannels) {
-    DVLOG(0) << "invalid channel count in extra data: " << extra_data->channels;
+    DLOG(ERROR) << "invalid channel count in extra data: "
+                << extra_data->channels;
     return false;
   }
 
@@ -223,7 +218,7 @@ static bool ParseOpusExtraData(const uint8* data, int data_size,
 
   if (!extra_data->channel_mapping) {
     if (extra_data->channels > kMaxChannelsWithDefaultLayout) {
-      DVLOG(0) << "Invalid extra data, missing stream map.";
+      DLOG(ERROR) << "Invalid extra data, missing stream map.";
       return false;
     }
 
@@ -234,8 +229,8 @@ static bool ParseOpusExtraData(const uint8* data, int data_size,
   }
 
   if (data_size < kOpusExtraDataStreamMapOffset + extra_data->channels) {
-    DVLOG(0) << "Invalid stream map; insufficient data for current channel "
-             << "count: " << extra_data->channels;
+    DLOG(ERROR) << "Invalid stream map; insufficient data for current channel "
+                << "count: " << extra_data->channels;
     return false;
   }
 
@@ -256,12 +251,20 @@ OpusAudioDecoder::OpusAudioDecoder(
       weak_factory_(this),
       demuxer_stream_(NULL),
       opus_decoder_(NULL),
-      bits_per_channel_(0),
       channel_layout_(CHANNEL_LAYOUT_NONE),
       samples_per_second_(0),
+      // Android uses a fixed point build of the opus decoder.
+#if defined(OS_ANDROID)
+      sample_format_(kSampleFormatS16),
+#else
+      sample_format_(kSampleFormatF32),
+#endif
+      bits_per_channel_(SampleFormatToBytesPerChannel(sample_format_) * 8),
       last_input_timestamp_(kNoTimestamp()),
       frames_to_discard_(0),
-      frame_delay_at_start_(0) {
+      frame_delay_at_start_(0),
+      first_output_frame_(kNoTimestamp()),
+      partial_frame_delay_at_start_(0) {
 }
 
 void OpusAudioDecoder::Initialize(
@@ -274,7 +277,7 @@ void OpusAudioDecoder::Initialize(
   if (demuxer_stream_) {
     // TODO(scherkus): initialization currently happens more than once in
     // PipelineIntegrationTest.BasicPlayback.
-    DVLOG(0) << "Initialize has already been called.";
+    DLOG(ERROR) << "Initialize has already been called.";
     CHECK(false);
   }
 
@@ -375,7 +378,7 @@ void OpusAudioDecoder::BufferReady(
   // occurs with some damaged files.
   if (input->timestamp() == kNoTimestamp() &&
       output_timestamp_helper_->base_timestamp() == kNoTimestamp()) {
-    DVLOG(1) << "Received a buffer without timestamps!";
+    DLOG(ERROR) << "Received a buffer without timestamps!";
     base::ResetAndReturn(&read_cb_).Run(kDecodeError, NULL);
     return;
   }
@@ -384,13 +387,22 @@ void OpusAudioDecoder::BufferReady(
       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->timestamp().InMicroseconds() << " us"
-             << " diff " << diff.InMicroseconds() << " us";
+    DLOG(ERROR) << "Input timestamps are not monotonically increasing! "
+                << " ts " << input->timestamp().InMicroseconds() << " us"
+                << " diff " << diff.InMicroseconds() << " us";
     base::ResetAndReturn(&read_cb_).Run(kDecodeError, NULL);
     return;
   }
 
+  // Skip samples should be equal to codec delay when the file starts and when
+  // there is a seek to zero.  Note: We're implicitly relying on FFmpeg to treat
+  // seek to zero as a seek to the actual first frame (which may be negative).
+  if (last_input_timestamp_ == kNoTimestamp() &&

[acolwell GONE FROM CHROMIUM, 2013/12/12 22:40:46]

This doesn't seem quite right. Shouldn't this be something like

if (start_input_timestamp_ == kNoTimestamp())
  start_input_timestamp_ = input->timestamp();

if (last_input_timestamp_ == kNoTimestamp() &&
    input->timestamp() == start_input_timestamp_) {
  frames_to_discard_ = frame_delay_at_start_;
}

I think if you do this you won't need  partial_frame_delay_at_start_  or the
(first_output_frame_ != kNoTimestamp()) and (first_output_frame_ ==
kNoTimestamp()) blocks in OpusAudioDecoder::Decode()

[DaleCurtis, 2013/12/12 23:36:04]

Done.  Required fixing AudioRendererImpl preroll!

+      input->timestamp() <= base::TimeDelta()) {
+    if (first_output_frame_ == kNoTimestamp())
+      frames_to_discard_ = frame_delay_at_start_;
+  }
+
   last_input_timestamp_ = input->timestamp();
 
   scoped_refptr<AudioBuffer> output_buffer;
@@ -414,53 +426,41 @@ bool OpusAudioDecoder::ConfigureDecoder() {
   const AudioDecoderConfig& config = demuxer_stream_->audio_decoder_config();
 
   if (config.codec() != kCodecOpus) {
-    DVLOG(0) << "codec must be kCodecOpus.";
+    DVLOG(1) << "Codec must be kCodecOpus.";
     return false;
   }
 
   const int channel_count =
       ChannelLayoutToChannelCount(config.channel_layout());
   if (!config.IsValidConfig() || channel_count > kMaxVorbisChannels) {
-    DVLOG(0) << "Invalid or unsupported audio stream -"
-             << " codec: " << config.codec()
-             << " channel count: " << channel_count
-             << " channel layout: " << config.channel_layout()
-             << " bits per channel: " << config.bits_per_channel()
-             << " samples per second: " << config.samples_per_second();
-    return false;
-  }
-
-  if (config.bits_per_channel() != kBitsPerChannel) {
-    DVLOG(0) << "16 bit samples required.";
+    DLOG(ERROR) << "Invalid or unsupported audio stream -"
+                << " codec: " << config.codec()
+                << " channel count: " << channel_count
+                << " channel layout: " << config.channel_layout()
+                << " bits per channel: " << config.bits_per_channel()
+                << " samples per second: " << config.samples_per_second();
     return false;
   }
 
   if (config.is_encrypted()) {
-    DVLOG(0) << "Encrypted audio stream not supported.";
+    DLOG(ERROR) << "Encrypted audio stream not supported.";
     return false;
   }
 
   if (opus_decoder_ &&
-      (bits_per_channel_ != config.bits_per_channel() ||
-       channel_layout_ != config.channel_layout() ||
+      (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();
+    DLOG(ERROR) << "Unsupported config change -"
+                << ", channel_layout: " << channel_layout_
+                << " -> " << config.channel_layout()
+                << ", sample_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 Extra Data.
   OpusExtraData opus_extra_data;
   if (!ParseOpusExtraData(config.extra_data(), config.extra_data_size(),
@@ -468,24 +468,23 @@ bool OpusAudioDecoder::ConfigureDecoder() {
                           &opus_extra_data))
     return false;
 
-  if (!config.codec_delay().InMicroseconds())
-    return false;
-
   // Convert from seconds to samples.
   timestamp_offset_ = config.codec_delay();
   frame_delay_at_start_ = TimeDeltaToAudioFrames(config.codec_delay(),
                                                  config.samples_per_second());
-  if (frame_delay_at_start_ < 0) {
-    DVLOG(1) << "Invalid file. Incorrect value for codec delay.";
+  if (timestamp_offset_ <= base::TimeDelta() || frame_delay_at_start_ < 0) {
+    DLOG(ERROR) << "Invalid file. Incorrect value for codec delay: "
+                << config.codec_delay().InMicroseconds();
     return false;
   }
+
   if (frame_delay_at_start_ != opus_extra_data.skip_samples) {
-    DVLOG(1) << "Invalid file. Codec Delay in container does not match the "
-             << "value in Opus Extra Data.";
+    DLOG(ERROR) << "Invalid file. Codec Delay in container does not match the "
+                << "value in Opus Extra Data.";
     return false;
   }
 
-  uint8 channel_mapping[kMaxVorbisChannels];
+  uint8 channel_mapping[kMaxVorbisChannels] = {0};
   memcpy(&channel_mapping,
          kDefaultOpusChannelLayout,
          kMaxChannelsWithDefaultLayout);
@@ -505,16 +504,17 @@ bool OpusAudioDecoder::ConfigureDecoder() {
                                                   channel_mapping,
                                                   &status);
   if (!opus_decoder_ || status != OPUS_OK) {
-    DVLOG(0) << "opus_multistream_decoder_create failed status="
-             << opus_strerror(status);
+    DLOG(ERROR) << "opus_multistream_decoder_create failed status="
+                << opus_strerror(status);
     return false;
   }
 
-  bits_per_channel_ = config.bits_per_channel();
   channel_layout_ = config.channel_layout();
   samples_per_second_ = config.samples_per_second();
   output_timestamp_helper_.reset(
       new AudioTimestampHelper(config.samples_per_second()));
+  first_output_frame_ = kNoTimestamp();
+  partial_frame_delay_at_start_ = 0;
   return true;
 }
 
@@ -535,68 +535,87 @@ void OpusAudioDecoder::ResetTimestampState() {
 
 bool OpusAudioDecoder::Decode(const scoped_refptr<DecoderBuffer>& input,
                               scoped_refptr<AudioBuffer>* output_buffer) {
-  int frames_decoded = opus_multistream_decode(opus_decoder_,
-                                               input->data(),
-                                               input->data_size(),
-                                               &output_buffer_[0],
-                                               kMaxOpusOutputPacketSizeSamples,
-                                               0);
+  // Allocate a buffer for the output samples.
+  *output_buffer = AudioBuffer::CreateBuffer(
+      sample_format_,
+      ChannelLayoutToChannelCount(channel_layout_),
+      kMaxOpusOutputPacketSizeSamples);
+  const int buffer_size =
+      output_buffer->get()->channel_count() *
+      output_buffer->get()->frame_count() *
+      SampleFormatToBytesPerChannel(sample_format_);
+
+  // Android uses a fixed point build of the opus decoder.
+#if defined(OS_ANDROID)
+  int16* int16_output_buffer = reinterpret_cast<int16*>(
+      output_buffer->get()->channel_data()[0]);
+  const int frames_decoded =
+      opus_multistream_decode(opus_decoder_,
+                              input->data(),
+                              input->data_size(),
+                              int16_output_buffer,
+                              buffer_size,
+                              0);
+#else
+  float* float_output_buffer = reinterpret_cast<float*>(
+      output_buffer->get()->channel_data()[0]);
+  const int frames_decoded =
+      opus_multistream_decode_float(opus_decoder_,
+                                    input->data(),
+                                    input->data_size(),
+                                    float_output_buffer,
+                                    buffer_size,
+                                    0);
+#endif
+
   if (frames_decoded < 0) {
-    DVLOG(0) << "opus_multistream_decode failed for"
-             << " timestamp: " << input->timestamp().InMicroseconds()
-             << " us, duration: " << input->duration().InMicroseconds()
-             << " us, packet size: " << input->data_size() << " bytes with"
-             << " status: " << opus_strerror(frames_decoded);
+    DLOG(ERROR) << "opus_multistream_decode failed for"
+                << " timestamp: " << input->timestamp().InMicroseconds()
+                << " us, duration: " << input->duration().InMicroseconds()
+                << " us, packet size: " << input->data_size() << " bytes with"
+                << " status: " << opus_strerror(frames_decoded);
     return false;
   }
 
-  uint8* decoded_audio_data = reinterpret_cast<uint8*>(&output_buffer_[0]);
-  int bytes_decoded = frames_decoded *
-      demuxer_stream_->audio_decoder_config().bytes_per_frame();
-  DCHECK_LE(bytes_decoded, kMaxOpusOutputPacketSizeBytes);
-
   if (output_timestamp_helper_->base_timestamp() == kNoTimestamp() &&
       !input->end_of_stream()) {
     DCHECK(input->timestamp() != kNoTimestamp());
     output_timestamp_helper_->SetBaseTimestamp(input->timestamp());
   }
 
-  // Skip samples should be equal to codec delay when the file starts and when
-  // there is a seek to zero.
-  // TODO(vigneshv): This should be checked for start of stream rather than
-  // input timestamp of zero to accomodate streams that don't start at zero.
-  if (input->timestamp() == base::TimeDelta())
-    frames_to_discard_ = frame_delay_at_start_;
-
-  if (bytes_decoded > 0 && frames_decoded > frames_to_discard_) {
-    // Copy the audio samples into an output buffer.
-    uint8* data[] = { decoded_audio_data };
-    *output_buffer = AudioBuffer::CopyFrom(
-        kSampleFormatS16,
-        ChannelLayoutToChannelCount(channel_layout_),
-        frames_decoded,
-        data,
-        output_timestamp_helper_->GetTimestamp() - timestamp_offset_,
-        output_timestamp_helper_->GetFrameDuration(frames_decoded));
-    output_timestamp_helper_->AddFrames(frames_decoded);
+  // Trim off any extraneous allocation.
+  DCHECK_LE(frames_decoded, output_buffer->get()->frame_count());
+  const int trim_frames = output_buffer->get()->frame_count() - frames_decoded;
+  if (trim_frames > 0)
+    output_buffer->get()->TrimEnd(trim_frames);
+
+  // Handle frame discard and trimming.
+  int frames_to_output = frames_decoded;
+  if (first_output_frame_ != kNoTimestamp()) {
+    if (input->timestamp() < first_output_frame_)
+      frames_to_discard_ = frames_decoded;
+    else if (input->timestamp() == first_output_frame_)
+      frames_to_discard_ = partial_frame_delay_at_start_;
+  }
+  if (frames_decoded > frames_to_discard_) {
     if (frames_to_discard_ > 0) {
       output_buffer->get()->TrimStart(frames_to_discard_);
-      frames_decoded -= frames_to_discard_;
+      frames_to_output -= frames_to_discard_;
       frames_to_discard_ = 0;
     }
     if (input->discard_padding().InMicroseconds() > 0) {
       int discard_padding = TimeDeltaToAudioFrames(input->discard_padding(),
                                                    samples_per_second_);
-      if (discard_padding < 0 || discard_padding > frames_decoded) {
+      if (discard_padding < 0 || discard_padding > frames_to_output) {
         DVLOG(1) << "Invalid file. Incorrect discard padding value.";
         return false;
       }
       output_buffer->get()->TrimEnd(discard_padding);
-      frames_decoded -= discard_padding;
+      frames_to_output -= discard_padding;
     }
-  } else if (bytes_decoded > 0) {
-    frames_to_discard_ -= frames_decoded;
-    frames_decoded = 0;
+  } else {
+    frames_to_discard_ -= frames_to_output;
+    frames_to_output = 0;
   }
 
   // Decoding finished successfully, update statistics.
@@ -604,9 +623,20 @@ bool OpusAudioDecoder::Decode(const scoped_refptr<DecoderBuffer>& input,
   statistics.audio_bytes_decoded = input->data_size();
   statistics_cb_.Run(statistics);
 
+  // Assign timestamp and duration to the buffer.
+  output_buffer->get()->set_timestamp(
+      output_timestamp_helper_->GetTimestamp() - timestamp_offset_);
+  output_buffer->get()->set_duration(
+      output_timestamp_helper_->GetFrameDuration(frames_to_output));
+  output_timestamp_helper_->AddFrames(frames_decoded);
+
   // Discard the buffer to indicate we need more data.
-  if (!frames_decoded)
+  if (!frames_to_output) {
     *output_buffer = NULL;
+  } else if (first_output_frame_ == kNoTimestamp()) {
+    first_output_frame_ = input->timestamp();
+    partial_frame_delay_at_start_ = frames_decoded - frames_to_output;
+  }
 
   return true;
 }