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Side by Side Diff: media/base/audio_buffer.cc

Issue 600143002: Adding new function ReadFrames() that returns the audio frame in planar (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Actually include the template function Created 6 years, 2 months ago
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1 // Copyright 2013 The Chromium Authors. All rights reserved. 1 // Copyright 2013 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/base/audio_buffer.h" 5 #include "media/base/audio_buffer.h"
6 6
7 #include "base/logging.h" 7 #include "base/logging.h"
8 #include "media/base/audio_bus.h" 8 #include "media/base/audio_bus.h"
9 #include "media/base/buffers.h" 9 #include "media/base/buffers.h"
10 #include "media/base/limits.h" 10 #include "media/base/limits.h"
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236 DCHECK(sample_format_ == kSampleFormatU8 || 236 DCHECK(sample_format_ == kSampleFormatU8 ||
237 sample_format_ == kSampleFormatS16 || 237 sample_format_ == kSampleFormatS16 ||
238 sample_format_ == kSampleFormatS32); 238 sample_format_ == kSampleFormatS32);
239 int bytes_per_channel = SampleFormatToBytesPerChannel(sample_format_); 239 int bytes_per_channel = SampleFormatToBytesPerChannel(sample_format_);
240 int frame_size = channel_count_ * bytes_per_channel; 240 int frame_size = channel_count_ * bytes_per_channel;
241 const uint8* source_data = data_.get() + source_frame_offset * frame_size; 241 const uint8* source_data = data_.get() + source_frame_offset * frame_size;
242 dest->FromInterleavedPartial( 242 dest->FromInterleavedPartial(
243 source_data, dest_frame_offset, frames_to_copy, bytes_per_channel); 243 source_data, dest_frame_offset, frames_to_copy, bytes_per_channel);
244 } 244 }
245 245
246 static inline int32 ConvertS16ToS32(int16 value) {
247 return static_cast<int32>(value) << 16;
248 }
249
250 static inline int32 ConvertF32ToS32(float value) {
251 return static_cast<int32>(value < 0
252 ? (-value) * std::numeric_limits<int32>::min()
253 : value * std::numeric_limits<int32>::max());
254 }
255
256 template <class Target, typename Converter>
257 void InterleaveToS32(const std::vector<uint8*>& channel_data,
258 size_t frames_to_copy,
259 int trim_start,
260 int32* dest_data,
261 Converter convert_func) {
262 for (size_t ch = 0; ch < channel_data.size(); ++ch) {
263 const Target* source_data =
264 reinterpret_cast<const Target*>(channel_data[ch]) + trim_start;
265 for (size_t i = 0, offset = ch; i < frames_to_copy;
266 ++i, offset += channel_data.size())
DaleCurtis 2014/09/29 23:01:32 {} since the for() spans multiple lines.
pwestin(chromium) 2014/09/30 00:15:17 Done.
267 dest_data[offset] = convert_func(source_data[i]);
268 }
269 }
270
271 void AudioBuffer::ReadFramesInterleavedS32(int frames_to_copy,
272 int32* dest_data) {
273 DCHECK_LE(frames_to_copy, adjusted_frame_count_);
274
275 switch (sample_format_) {
276 case kSampleFormatU8:
277 NOTIMPLEMENTED();
278 break;
279 case kSampleFormatS16:
280 // Format is interleaved signed16. Convert each value into int32 and
281 // insert into output channel data.
282 InterleaveToS32<int16>(channel_data_,
283 frames_to_copy * channel_count_,
284 trim_start_,
285 dest_data,
286 ConvertS16ToS32);
287 break;
288 case kSampleFormatS32: {
289 // Format is interleaved signed32; just copy the data.
290 const int32* source_data =
291 reinterpret_cast<const int32*>(channel_data_[0]) + trim_start_;
292 memcpy(dest_data,
293 source_data,
294 frames_to_copy * channel_count_ * sizeof(int32));
295 } break;
296 case kSampleFormatF32:
297 // Format is interleaved float. Convert each value into int32 and insert
298 // into output channel data.
299 InterleaveToS32<float>(channel_data_,
300 frames_to_copy * channel_count_,
301 trim_start_,
302 dest_data,
303 ConvertF32ToS32);
304 break;
305 case kSampleFormatPlanarS16:
306 // Format is planar signed 16 bit. Convert each value into int32 and
307 // insert into output channel data.
308 InterleaveToS32<int16>(channel_data_,
309 frames_to_copy,
310 trim_start_,
311 dest_data,
312 ConvertS16ToS32);
313 break;
314 case kSampleFormatPlanarF32:
315 // Format is planar float. Convert each value into int32 and insert into
316 // output channel data.
317 InterleaveToS32<float>(channel_data_,
318 frames_to_copy,
319 trim_start_,
320 dest_data,
321 ConvertF32ToS32);
322 break;
323 case kUnknownSampleFormat:
324 NOTREACHED();
325 break;
326 }
327 }
328
246 void AudioBuffer::TrimStart(int frames_to_trim) { 329 void AudioBuffer::TrimStart(int frames_to_trim) {
247 CHECK_GE(frames_to_trim, 0); 330 CHECK_GE(frames_to_trim, 0);
248 CHECK_LE(frames_to_trim, adjusted_frame_count_); 331 CHECK_LE(frames_to_trim, adjusted_frame_count_);
249 332
250 // Adjust the number of frames in this buffer and where the start really is. 333 // Adjust the number of frames in this buffer and where the start really is.
251 adjusted_frame_count_ -= frames_to_trim; 334 adjusted_frame_count_ -= frames_to_trim;
252 trim_start_ += frames_to_trim; 335 trim_start_ += frames_to_trim;
253 336
254 // Adjust timestamp_ and duration_ to reflect the smaller number of frames. 337 // Adjust timestamp_ and duration_ to reflect the smaller number of frames.
255 const base::TimeDelta old_duration = duration_; 338 const base::TimeDelta old_duration = duration_;
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303 } 386 }
304 } else { 387 } else {
305 CHECK_EQ(frames_to_copy, 0); 388 CHECK_EQ(frames_to_copy, 0);
306 } 389 }
307 390
308 // Trim the leftover data off the end of the buffer and update duration. 391 // Trim the leftover data off the end of the buffer and update duration.
309 TrimEnd(frames_to_trim); 392 TrimEnd(frames_to_trim);
310 } 393 }
311 394
312 } // namespace media 395 } // namespace media
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