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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" |
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156 int frames_read = av_frame->nb_samples; | 156 int frames_read = av_frame->nb_samples; |
157 if (frames_read < 0) { | 157 if (frames_read < 0) { |
158 continue_decoding = false; | 158 continue_decoding = false; |
159 break; | 159 break; |
160 } | 160 } |
161 | 161 |
162 // Truncate, if necessary, if the destination isn't big enough. | 162 // Truncate, if necessary, if the destination isn't big enough. |
163 if (current_frame + frames_read > audio_bus->frames()) | 163 if (current_frame + frames_read > audio_bus->frames()) |
164 frames_read = audio_bus->frames() - current_frame; | 164 frames_read = audio_bus->frames() - current_frame; |
165 | 165 |
166 // Deinterleave each channel and convert to 32bit floating-point | 166 // Deinterleave each channel and convert to 32bit floating-point with |
167 // with nominal range -1.0 -> +1.0. | 167 // nominal range -1.0 -> +1.0. If the output is already in float planar |
168 audio_bus->FromInterleavedPartial( | 168 // format, just copy it into the AudioBus. |
169 av_frame->data[0], current_frame, frames_read, bytes_per_sample); | 169 if (codec_context_->sample_fmt == AV_SAMPLE_FMT_FLT) { |
| 170 float* decoded_audio_data = reinterpret_cast<float*>(av_frame->data[0]); |
| 171 int channels = audio_bus->channels(); |
| 172 for (int ch = 0; ch < channels; ++ch) { |
| 173 float* bus_data = audio_bus->channel(ch) + current_frame; |
| 174 for (int i = 0, offset = ch; i < frames_read; |
| 175 ++i, offset += channels) { |
| 176 bus_data[i] = decoded_audio_data[offset]; |
| 177 } |
| 178 } |
| 179 } else if (codec_context_->sample_fmt == AV_SAMPLE_FMT_FLTP) { |
| 180 for (int ch = 0; ch < audio_bus->channels(); ++ch) { |
| 181 memcpy(audio_bus->channel(ch) + current_frame, |
| 182 av_frame->extended_data[ch], sizeof(float) * frames_read); |
| 183 } |
| 184 } else { |
| 185 audio_bus->FromInterleavedPartial( |
| 186 av_frame->data[0], current_frame, frames_read, bytes_per_sample); |
| 187 } |
170 | 188 |
171 current_frame += frames_read; | 189 current_frame += frames_read; |
172 } while (packet_temp.size > 0); | 190 } while (packet_temp.size > 0); |
173 av_free_packet(&packet); | 191 av_free_packet(&packet); |
174 } | 192 } |
175 | 193 |
176 // Zero any remaining frames. | 194 // Zero any remaining frames. |
177 audio_bus->ZeroFramesPartial( | 195 audio_bus->ZeroFramesPartial( |
178 current_frame, audio_bus->frames() - current_frame); | 196 current_frame, audio_bus->frames() - current_frame); |
179 | 197 |
180 // Returns the actual number of sample-frames decoded. | 198 // Returns the actual number of sample-frames decoded. |
181 // Ideally this represents the "true" exact length of the file. | 199 // Ideally this represents the "true" exact length of the file. |
182 return current_frame; | 200 return current_frame; |
183 } | 201 } |
184 | 202 |
185 } // namespace media | 203 } // namespace media |
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