Wire up codec_delay() to MP3StreamParser and FFmpegAudioDecoder.

media/filters/ffmpeg_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/ffmpeg_audio_decoder.h"
 
 #include "base/callback_helpers.h"
 #include "base/single_thread_task_runner.h"
 #include "media/base/audio_buffer.h"
 #include "media/base/audio_bus.h"
@@ skipping 244 matching lines @@
   // Make sure we are notified if http://crbug.com/49709 returns.  Issue also
   // occurs with some damaged files.
   if (!buffer->end_of_stream() && buffer->timestamp() == kNoTimestamp() &&
       output_timestamp_helper_->base_timestamp() == kNoTimestamp()) {
     DVLOG(1) << "Received a buffer without timestamps!";
     decode_cb.Run(kDecodeError, NULL);
     return;
   }
 
   if (!buffer->end_of_stream()) {
-    if (last_input_timestamp_ == kNoTimestamp() &&
-        codec_context_->codec_id == AV_CODEC_ID_VORBIS &&
+    DCHECK(buffer->timestamp() != kNoTimestamp());
+    const bool first_buffer =
+        last_input_timestamp_ == kNoTimestamp() &&
+        output_timestamp_helper_->base_timestamp() == kNoTimestamp();
+    if (first_buffer && codec_context_->codec_id == AV_CODEC_ID_VORBIS &&
         buffer->timestamp() < base::TimeDelta()) {
       // Dropping frames for negative timestamps as outlined in section A.2
       // in the Vorbis spec. http://xiph.org/vorbis/doc/Vorbis_I_spec.html
-      output_frames_to_drop_ = floor(0.5 + -buffer->timestamp().InSecondsF() *
-                                               config_.samples_per_second());
+      DCHECK_EQ(output_frames_to_drop_, 0);
+      output_frames_to_drop_ =
+          0.5 +
+          -buffer->timestamp().InSecondsF() * config_.samples_per_second();
+
+      // If we are dropping samples for Vorbis, the timeline always starts at 0.
+      output_timestamp_helper_->SetBaseTimestamp(base::TimeDelta());
     } else {
-      if (last_input_timestamp_ != kNoTimestamp() &&
-          buffer->timestamp() < last_input_timestamp_) {
+      if (first_buffer) {
+        output_timestamp_helper_->SetBaseTimestamp(buffer->timestamp());
+      } else if (buffer->timestamp() < last_input_timestamp_) {
         const base::TimeDelta diff =
             buffer->timestamp() - last_input_timestamp_;
-        DLOG(WARNING)
-            << "Input timestamps are not monotonically increasing! "
-            << " ts " << buffer->timestamp().InMicroseconds() << " us"
-            << " diff " << diff.InMicroseconds() << " us";
+        DLOG(WARNING) << "Input timestamps are not monotonically increasing! "
+                      << " ts " << buffer->timestamp().InMicroseconds() << " us"
+                      << " diff " << diff.InMicroseconds() << " us";
       }
 
       last_input_timestamp_ = buffer->timestamp();
     }
   }
 
   // Transition to kFlushCodec on the first end of stream buffer.
   if (state_ == kNormal && buffer->end_of_stream()) {
     state_ = kFlushCodec;
   }
@@ skipping 57 matching lines @@
           << buffer->data_size() << " bytes";
 
       break;
     }
 
     // Update packet size and data pointer in case we need to call the decoder
     // with the remaining bytes from this packet.
     packet.size -= result;
     packet.data += result;
 
-    if (output_timestamp_helper_->base_timestamp() == kNoTimestamp() &&
-        !buffer->end_of_stream()) {
-      DCHECK(buffer->timestamp() != kNoTimestamp());
-      if (output_frames_to_drop_ > 0) {
-        // Currently Vorbis is the only codec that causes us to drop samples.
-        // If we have to drop samples it always means the timeline starts at 0.
-        DCHECK_EQ(codec_context_->codec_id, AV_CODEC_ID_VORBIS);
-        output_timestamp_helper_->SetBaseTimestamp(base::TimeDelta());
-      } else {
-        output_timestamp_helper_->SetBaseTimestamp(buffer->timestamp());
-      }
-    }
-
     scoped_refptr<AudioBuffer> output;
     int decoded_frames = 0;
     int original_frames = 0;
     int channels = DetermineChannels(av_frame_.get());
     if (frame_decoded) {
       if (av_frame_->sample_rate != config_.samples_per_second() ||
           channels != ChannelLayoutToChannelCount(config_.channel_layout()) ||
           av_frame_->format != av_sample_format_) {
         DLOG(ERROR) << "Unsupported midstream configuration change!"
                     << " Sample Rate: " << av_frame_->sample_rate << " vs "
@@ skipping 95 matching lines @@
                 << codec_context_->codec_id;
     ReleaseFFmpegResources();
     state_ = kUninitialized;
     return false;
   }
 
   // Success!
   av_frame_.reset(av_frame_alloc());
   output_timestamp_helper_.reset(
       new AudioTimestampHelper(config_.samples_per_second()));
+  ResetTimestampState();
 
   av_sample_format_ = codec_context_->sample_fmt;
 
   if (codec_context_->channels !=
       ChannelLayoutToChannelCount(config_.channel_layout())) {
     DLOG(ERROR) << "Audio configuration specified "
                 << ChannelLayoutToChannelCount(config_.channel_layout())
                 << " channels, but FFmpeg thinks the file contains "
                 << codec_context_->channels << " channels";
     ReleaseFFmpegResources();
     state_ = kUninitialized;
     return false;
   }
+
+  output_frames_to_drop_ = config_.codec_delay();
   return true;
 }
 
 void FFmpegAudioDecoder::ResetTimestampState() {
   output_timestamp_helper_->SetBaseTimestamp(kNoTimestamp());
   last_input_timestamp_ = kNoTimestamp();
   output_frames_to_drop_ = 0;
 }
 
 }  // namespace media