Chromium Code Reviews| OLD | NEW |
|---|---|
| 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. |
| 2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
| 4 | 4 |
| 5 #include "media/filters/audio_file_reader.h" | 5 #include "media/filters/audio_file_reader.h" |
| 6 | 6 |
| 7 #include "base/logging.h" | 7 #include "base/logging.h" |
| 8 #include "base/time.h" | 8 #include "base/time.h" |
| 9 #include "media/base/audio_bus.h" | 9 #include "media/base/audio_bus.h" |
| 10 #include "media/ffmpeg/ffmpeg_common.h" | 10 #include "media/ffmpeg/ffmpeg_common.h" |
| 11 #include "media/filters/ffmpeg_glue.h" | 11 #include "media/filters/ffmpeg_glue.h" |
| 12 | 12 |
| 13 namespace media { | 13 namespace media { |
| 14 | 14 |
| 15 AudioFileReader::AudioFileReader(FFmpegURLProtocol* protocol) | 15 AudioFileReader::AudioFileReader(FFmpegURLProtocol* protocol) |
| 16 : codec_context_(NULL), | 16 : codec_context_(NULL), |
| 17 stream_index_(0), | 17 stream_index_(0), |
| 18 protocol_(protocol) { | 18 protocol_(protocol), |
| 19 channels_(0), | |
| 20 sample_rate_(0), | |
| 21 av_sample_format_(0) { | |
| 19 } | 22 } |
| 20 | 23 |
| 21 AudioFileReader::~AudioFileReader() { | 24 AudioFileReader::~AudioFileReader() { |
| 22 Close(); | 25 Close(); |
| 23 } | 26 } |
| 24 | 27 |
| 25 int AudioFileReader::channels() const { | |
| 26 return codec_context_->channels; | |
| 27 } | |
| 28 | |
| 29 int AudioFileReader::sample_rate() const { | |
| 30 return codec_context_->sample_rate; | |
| 31 } | |
| 32 | |
| 33 base::TimeDelta AudioFileReader::duration() const { | 28 base::TimeDelta AudioFileReader::duration() const { |
| 34 const AVRational av_time_base = {1, AV_TIME_BASE}; | 29 const AVRational av_time_base = {1, AV_TIME_BASE}; |
| 35 | 30 |
| 36 // Add one microsecond to avoid rounding-down errors which can occur when | 31 // Add one microsecond to avoid rounding-down errors which can occur when |
| 37 // |duration| has been calculated from an exact number of sample-frames. | 32 // |duration| has been calculated from an exact number of sample-frames. |
| 38 // One microsecond is much less than the time of a single sample-frame | 33 // One microsecond is much less than the time of a single sample-frame |
| 39 // at any real-world sample-rate. | 34 // at any real-world sample-rate. |
| 40 return ConvertFromTimeBase( | 35 return ConvertFromTimeBase( |
| 41 av_time_base, glue_->format_context()->duration + 1); | 36 av_time_base, glue_->format_context()->duration + 1); |
| 42 } | 37 } |
| (...skipping 60 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 103 } | 98 } |
| 104 | 99 |
| 105 // Verify the channel layout is supported by Chrome. Acts as a sanity check | 100 // Verify the channel layout is supported by Chrome. Acts as a sanity check |
| 106 // against invalid files. See http://crbug.com/171962 | 101 // against invalid files. See http://crbug.com/171962 |
| 107 if (ChannelLayoutToChromeChannelLayout( | 102 if (ChannelLayoutToChromeChannelLayout( |
| 108 codec_context_->channel_layout, codec_context_->channels) == | 103 codec_context_->channel_layout, codec_context_->channels) == |
| 109 CHANNEL_LAYOUT_UNSUPPORTED) { | 104 CHANNEL_LAYOUT_UNSUPPORTED) { |
| 110 return false; | 105 return false; |
| 111 } | 106 } |
| 112 | 107 |
| 108 // Store initial values to guard against mid-frame configuration changes. | |
| 109 channels_ = codec_context_->channels; | |
| 110 sample_rate_ = codec_context_->sample_rate; | |
| 111 av_sample_format_ = codec_context_->sample_fmt; | |
| 112 | |
| 113 return true; | 113 return true; |
| 114 } | 114 } |
| 115 | 115 |
| 116 void AudioFileReader::Close() { | 116 void AudioFileReader::Close() { |
| 117 if (codec_context_) { | 117 if (codec_context_) { |
| 118 avcodec_close(codec_context_); | 118 avcodec_close(codec_context_); |
| 119 codec_context_ = NULL; | 119 codec_context_ = NULL; |
| 120 } | 120 } |
| 121 } | 121 } |
| 122 | 122 |
| (...skipping 49 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 172 if (!frame_decoded) | 172 if (!frame_decoded) |
| 173 continue; | 173 continue; |
| 174 | 174 |
| 175 // Determine the number of sample-frames we just decoded. Check overflow. | 175 // Determine the number of sample-frames we just decoded. Check overflow. |
| 176 int frames_read = av_frame->nb_samples; | 176 int frames_read = av_frame->nb_samples; |
| 177 if (frames_read < 0) { | 177 if (frames_read < 0) { |
| 178 continue_decoding = false; | 178 continue_decoding = false; |
| 179 break; | 179 break; |
| 180 } | 180 } |
| 181 | 181 |
| 182 if (av_frame->sample_rate != sample_rate_ || | |
| 183 av_frame->channels != channels_ || | |
| 184 av_frame->format != av_sample_format_) { | |
| 185 DLOG(ERROR) << "Unsupported mid-frame configuration change!" | |
|
scherkus (not reviewing)
2013/02/12 02:16:56
consistent terminology nit: s/mid-frame/mid-stream
DaleCurtis
2013/02/12 02:41:24
Done.
| |
| 186 << " Sample Rate: " << av_frame->sample_rate << " vs " | |
| 187 << sample_rate_ | |
| 188 << ", Channels: " << av_frame->channels << " vs " | |
| 189 << channels_ | |
| 190 << ", Sample Format: " << av_frame->format << " vs " | |
| 191 << av_sample_format_; | |
| 192 | |
| 193 // This is an unrecoverable error, so bail out. | |
| 194 continue_decoding = false; | |
| 195 break; | |
| 196 } | |
| 197 | |
| 182 // Truncate, if necessary, if the destination isn't big enough. | 198 // Truncate, if necessary, if the destination isn't big enough. |
| 183 if (current_frame + frames_read > audio_bus->frames()) | 199 if (current_frame + frames_read > audio_bus->frames()) |
| 184 frames_read = audio_bus->frames() - current_frame; | 200 frames_read = audio_bus->frames() - current_frame; |
| 185 | 201 |
| 186 // Deinterleave each channel and convert to 32bit floating-point with | 202 // Deinterleave each channel and convert to 32bit floating-point with |
| 187 // nominal range -1.0 -> +1.0. If the output is already in float planar | 203 // nominal range -1.0 -> +1.0. If the output is already in float planar |
| 188 // format, just copy it into the AudioBus. | 204 // format, just copy it into the AudioBus. |
| 189 if (codec_context_->sample_fmt == AV_SAMPLE_FMT_FLT) { | 205 if (codec_context_->sample_fmt == AV_SAMPLE_FMT_FLT) { |
| 190 float* decoded_audio_data = reinterpret_cast<float*>(av_frame->data[0]); | 206 float* decoded_audio_data = reinterpret_cast<float*>(av_frame->data[0]); |
| 191 int channels = audio_bus->channels(); | 207 int channels = audio_bus->channels(); |
| (...skipping 22 matching lines...) Expand all Loading... | |
| 214 // Zero any remaining frames. | 230 // Zero any remaining frames. |
| 215 audio_bus->ZeroFramesPartial( | 231 audio_bus->ZeroFramesPartial( |
| 216 current_frame, audio_bus->frames() - current_frame); | 232 current_frame, audio_bus->frames() - current_frame); |
| 217 | 233 |
| 218 // Returns the actual number of sample-frames decoded. | 234 // Returns the actual number of sample-frames decoded. |
| 219 // Ideally this represents the "true" exact length of the file. | 235 // Ideally this represents the "true" exact length of the file. |
| 220 return current_frame; | 236 return current_frame; |
| 221 } | 237 } |
| 222 | 238 |
| 223 } // namespace media | 239 } // namespace media |
| OLD | NEW |