1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
|
// 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"
#include "media/base/audio_decoder_config.h"
#include "media/base/audio_discard_helper.h"
#include "media/base/bind_to_current_loop.h"
#include "media/base/decoder_buffer.h"
#include "media/base/limits.h"
#include "media/base/timestamp_constants.h"
#include "media/ffmpeg/ffmpeg_common.h"
#include "media/filters/ffmpeg_glue.h"
namespace media {
// Returns true if the decode result was end of stream.
static inline bool IsEndOfStream(int result,
int decoded_size,
const scoped_refptr<DecoderBuffer>& input) {
// Three conditions to meet to declare end of stream for this decoder:
// 1. FFmpeg didn't read anything.
// 2. FFmpeg didn't output anything.
// 3. An end of stream buffer is received.
return result == 0 && decoded_size == 0 && input->end_of_stream();
}
// Return the number of channels from the data in |frame|.
static inline int DetermineChannels(AVFrame* frame) {
#if defined(CHROMIUM_NO_AVFRAME_CHANNELS)
// When use_system_ffmpeg==1, libav's AVFrame doesn't have channels field.
return av_get_channel_layout_nb_channels(frame->channel_layout);
#else
return frame->channels;
#endif
}
// Called by FFmpeg's allocation routine to free a buffer. |opaque| is the
// AudioBuffer allocated, so unref it.
static void ReleaseAudioBufferImpl(void* opaque, uint8* data) {
scoped_refptr<AudioBuffer> buffer;
buffer.swap(reinterpret_cast<AudioBuffer**>(&opaque));
}
// Called by FFmpeg's allocation routine to allocate a buffer. Uses
// AVCodecContext.opaque to get the object reference in order to call
// GetAudioBuffer() to do the actual allocation.
static int GetAudioBuffer(struct AVCodecContext* s, AVFrame* frame, int flags) {
DCHECK(s->codec->capabilities & CODEC_CAP_DR1);
DCHECK_EQ(s->codec_type, AVMEDIA_TYPE_AUDIO);
// Since this routine is called by FFmpeg when a buffer is required for audio
// data, use the values supplied by FFmpeg (ignoring the current settings).
// FFmpegDecode() gets to determine if the buffer is useable or not.
AVSampleFormat format = static_cast<AVSampleFormat>(frame->format);
SampleFormat sample_format = AVSampleFormatToSampleFormat(format);
int channels = DetermineChannels(frame);
if (channels <= 0 || channels >= limits::kMaxChannels) {
DLOG(ERROR) << "Requested number of channels (" << channels
<< ") exceeds limit.";
return AVERROR(EINVAL);
}
int bytes_per_channel = SampleFormatToBytesPerChannel(sample_format);
if (frame->nb_samples <= 0)
return AVERROR(EINVAL);
if (s->channels != channels) {
DLOG(ERROR) << "AVCodecContext and AVFrame disagree on channel count.";
return AVERROR(EINVAL);
}
// Determine how big the buffer should be and allocate it. FFmpeg may adjust
// how big each channel data is in order to meet the alignment policy, so
// we need to take this into consideration.
int buffer_size_in_bytes =
av_samples_get_buffer_size(&frame->linesize[0],
channels,
frame->nb_samples,
format,
AudioBuffer::kChannelAlignment);
// Check for errors from av_samples_get_buffer_size().
if (buffer_size_in_bytes < 0)
return buffer_size_in_bytes;
int frames_required = buffer_size_in_bytes / bytes_per_channel / channels;
DCHECK_GE(frames_required, frame->nb_samples);
scoped_refptr<AudioBuffer> buffer = AudioBuffer::CreateBuffer(
sample_format,
ChannelLayoutToChromeChannelLayout(s->channel_layout, s->channels),
channels,
s->sample_rate,
frames_required);
// Initialize the data[] and extended_data[] fields to point into the memory
// allocated for AudioBuffer. |number_of_planes| will be 1 for interleaved
// audio and equal to |channels| for planar audio.
int number_of_planes = buffer->channel_data().size();
if (number_of_planes <= AV_NUM_DATA_POINTERS) {
DCHECK_EQ(frame->extended_data, frame->data);
for (int i = 0; i < number_of_planes; ++i)
frame->data[i] = buffer->channel_data()[i];
} else {
// There are more channels than can fit into data[], so allocate
// extended_data[] and fill appropriately.
frame->extended_data = static_cast<uint8**>(
av_malloc(number_of_planes * sizeof(*frame->extended_data)));
int i = 0;
for (; i < AV_NUM_DATA_POINTERS; ++i)
frame->extended_data[i] = frame->data[i] = buffer->channel_data()[i];
for (; i < number_of_planes; ++i)
frame->extended_data[i] = buffer->channel_data()[i];
}
// Now create an AVBufferRef for the data just allocated. It will own the
// reference to the AudioBuffer object.
void* opaque = NULL;
buffer.swap(reinterpret_cast<AudioBuffer**>(&opaque));
frame->buf[0] = av_buffer_create(
frame->data[0], buffer_size_in_bytes, ReleaseAudioBufferImpl, opaque, 0);
return 0;
}
FFmpegAudioDecoder::FFmpegAudioDecoder(
const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
const scoped_refptr<MediaLog>& media_log)
: task_runner_(task_runner),
state_(kUninitialized),
av_sample_format_(0),
media_log_(media_log) {
}
FFmpegAudioDecoder::~FFmpegAudioDecoder() {
DCHECK(task_runner_->BelongsToCurrentThread());
if (state_ != kUninitialized)
ReleaseFFmpegResources();
}
std::string FFmpegAudioDecoder::GetDisplayName() const {
return "FFmpegAudioDecoder";
}
void FFmpegAudioDecoder::Initialize(const AudioDecoderConfig& config,
const InitCB& init_cb,
const OutputCB& output_cb) {
DCHECK(task_runner_->BelongsToCurrentThread());
DCHECK(!config.is_encrypted());
FFmpegGlue::InitializeFFmpeg();
config_ = config;
InitCB bound_init_cb = BindToCurrentLoop(init_cb);
if (!config.IsValidConfig() || !ConfigureDecoder()) {
bound_init_cb.Run(false);
return;
}
// Success!
output_cb_ = BindToCurrentLoop(output_cb);
state_ = kNormal;
bound_init_cb.Run(true);
}
void FFmpegAudioDecoder::Decode(const scoped_refptr<DecoderBuffer>& buffer,
const DecodeCB& decode_cb) {
DCHECK(task_runner_->BelongsToCurrentThread());
DCHECK(!decode_cb.is_null());
CHECK_NE(state_, kUninitialized);
DecodeCB decode_cb_bound = BindToCurrentLoop(decode_cb);
if (state_ == kError) {
decode_cb_bound.Run(kDecodeError);
return;
}
// Do nothing if decoding has finished.
if (state_ == kDecodeFinished) {
decode_cb_bound.Run(kOk);
return;
}
DecodeBuffer(buffer, decode_cb_bound);
}
void FFmpegAudioDecoder::Reset(const base::Closure& closure) {
DCHECK(task_runner_->BelongsToCurrentThread());
avcodec_flush_buffers(codec_context_.get());
state_ = kNormal;
ResetTimestampState();
task_runner_->PostTask(FROM_HERE, closure);
}
void FFmpegAudioDecoder::DecodeBuffer(
const scoped_refptr<DecoderBuffer>& buffer,
const DecodeCB& decode_cb) {
DCHECK(task_runner_->BelongsToCurrentThread());
DCHECK_NE(state_, kUninitialized);
DCHECK_NE(state_, kDecodeFinished);
DCHECK_NE(state_, kError);
DCHECK(buffer.get());
// 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()) {
DVLOG(1) << "Received a buffer without timestamps!";
decode_cb.Run(kDecodeError);
return;
}
bool has_produced_frame;
do {
has_produced_frame = false;
if (!FFmpegDecode(buffer, &has_produced_frame)) {
state_ = kError;
decode_cb.Run(kDecodeError);
return;
}
// Repeat to flush the decoder after receiving EOS buffer.
} while (buffer->end_of_stream() && has_produced_frame);
if (buffer->end_of_stream())
state_ = kDecodeFinished;
decode_cb.Run(kOk);
}
bool FFmpegAudioDecoder::FFmpegDecode(
const scoped_refptr<DecoderBuffer>& buffer,
bool* has_produced_frame) {
DCHECK(!*has_produced_frame);
AVPacket packet;
av_init_packet(&packet);
if (buffer->end_of_stream()) {
packet.data = NULL;
packet.size = 0;
} else {
packet.data = const_cast<uint8*>(buffer->data());
packet.size = buffer->data_size();
}
// Each audio packet may contain several frames, so we must call the decoder
// until we've exhausted the packet. Regardless of the packet size we always
// want to hand it to the decoder at least once, otherwise we would end up
// skipping end of stream packets since they have a size of zero.
do {
int frame_decoded = 0;
const int result = avcodec_decode_audio4(
codec_context_.get(), av_frame_.get(), &frame_decoded, &packet);
if (result < 0) {
DCHECK(!buffer->end_of_stream())
<< "End of stream buffer produced an error! "
<< "This is quite possibly a bug in the audio decoder not handling "
<< "end of stream AVPackets correctly.";
MEDIA_LOG(DEBUG, media_log_)
<< "Dropping audio frame which failed decode with timestamp: "
<< buffer->timestamp().InMicroseconds()
<< " us, duration: " << buffer->duration().InMicroseconds()
<< " us, packet size: " << 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;
scoped_refptr<AudioBuffer> output;
const 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 "
<< config_.samples_per_second()
<< ", Channels: " << channels << " vs "
<< ChannelLayoutToChannelCount(config_.channel_layout())
<< ", Sample Format: " << av_frame_->format << " vs "
<< av_sample_format_;
if (config_.codec() == kCodecAAC &&
av_frame_->sample_rate == 2 * config_.samples_per_second()) {
MEDIA_LOG(DEBUG, media_log_)
<< "Implicit HE-AAC signalling is being"
<< " used. Please use mp4a.40.5 instead of"
<< " mp4a.40.2 in the mimetype.";
}
// This is an unrecoverable error, so bail out.
av_frame_unref(av_frame_.get());
return false;
}
// Get the AudioBuffer that the data was decoded into. Adjust the number
// of frames, in case fewer than requested were actually decoded.
output = reinterpret_cast<AudioBuffer*>(
av_buffer_get_opaque(av_frame_->buf[0]));
DCHECK_EQ(ChannelLayoutToChannelCount(config_.channel_layout()),
output->channel_count());
const int unread_frames = output->frame_count() - av_frame_->nb_samples;
DCHECK_GE(unread_frames, 0);
if (unread_frames > 0)
output->TrimEnd(unread_frames);
av_frame_unref(av_frame_.get());
}
// WARNING: |av_frame_| no longer has valid data at this point.
const int decoded_frames = frame_decoded ? output->frame_count() : 0;
if (IsEndOfStream(result, decoded_frames, buffer)) {
DCHECK_EQ(packet.size, 0);
} else if (discard_helper_->ProcessBuffers(buffer, output)) {
*has_produced_frame = true;
output_cb_.Run(output);
}
} while (packet.size > 0);
return true;
}
void FFmpegAudioDecoder::ReleaseFFmpegResources() {
codec_context_.reset();
av_frame_.reset();
}
bool FFmpegAudioDecoder::ConfigureDecoder() {
if (!config_.IsValidConfig()) {
DLOG(ERROR) << "Invalid audio stream -"
<< " codec: " << config_.codec()
<< " 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()) {
DLOG(ERROR) << "Encrypted audio stream not supported";
return false;
}
// Release existing decoder resources if necessary.
ReleaseFFmpegResources();
// Initialize AVCodecContext structure.
codec_context_.reset(avcodec_alloc_context3(NULL));
AudioDecoderConfigToAVCodecContext(config_, codec_context_.get());
codec_context_->opaque = this;
codec_context_->get_buffer2 = GetAudioBuffer;
codec_context_->refcounted_frames = 1;
AVCodec* codec = avcodec_find_decoder(codec_context_->codec_id);
if (!codec || avcodec_open2(codec_context_.get(), codec, NULL) < 0) {
DLOG(ERROR) << "Could not initialize audio decoder: "
<< codec_context_->codec_id;
ReleaseFFmpegResources();
state_ = kUninitialized;
return false;
}
// Success!
av_frame_.reset(av_frame_alloc());
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;
}
ResetTimestampState();
return true;
}
void FFmpegAudioDecoder::ResetTimestampState() {
discard_helper_.reset(new AudioDiscardHelper(config_.samples_per_second(),
config_.codec_delay()));
discard_helper_->Reset(config_.codec_delay());
}
} // namespace media
|