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
3 // found in the LICENSE file.
5 #include "media/filters/audio_file_reader.h"
7 #include "base/logging.h"
9 #include "media/base/audio_bus.h"
10 #include "media/ffmpeg/ffmpeg_common.h"
11 #include "media/filters/ffmpeg_glue.h"
15 AudioFileReader::AudioFileReader(FFmpegURLProtocol
* protocol
)
16 : codec_context_(NULL
),
21 av_sample_format_(0) {
24 AudioFileReader::~AudioFileReader() {
28 base::TimeDelta
AudioFileReader::duration() const {
29 const AVRational av_time_base
= {1, AV_TIME_BASE
};
31 // Add one microsecond to avoid rounding-down errors which can occur when
32 // |duration| has been calculated from an exact number of sample-frames.
33 // One microsecond is much less than the time of a single sample-frame
34 // at any real-world sample-rate.
35 return ConvertFromTimeBase(
36 av_time_base
, glue_
->format_context()->duration
+ 1);
39 int64
AudioFileReader::number_of_frames() const {
40 return static_cast<int64
>(duration().InSecondsF() * sample_rate());
43 bool AudioFileReader::Open() {
44 glue_
.reset(new FFmpegGlue(protocol_
));
45 AVFormatContext
* format_context
= glue_
->format_context();
47 // Open FFmpeg AVFormatContext.
48 if (!glue_
->OpenContext()) {
49 DLOG(WARNING
) << "AudioFileReader::Open() : error in avformat_open_input()";
53 // Get the codec context.
54 codec_context_
= NULL
;
55 for (size_t i
= 0; i
< format_context
->nb_streams
; ++i
) {
56 AVCodecContext
* c
= format_context
->streams
[i
]->codec
;
57 if (c
->codec_type
== AVMEDIA_TYPE_AUDIO
) {
68 int result
= avformat_find_stream_info(format_context
, NULL
);
71 << "AudioFileReader::Open() : error in avformat_find_stream_info()";
75 AVCodec
* codec
= avcodec_find_decoder(codec_context_
->codec_id
);
77 // MP3 decodes to S16P which we don't support, tell it to use S16 instead.
78 if (codec_context_
->sample_fmt
== AV_SAMPLE_FMT_S16P
)
79 codec_context_
->request_sample_fmt
= AV_SAMPLE_FMT_S16
;
81 if ((result
= avcodec_open2(codec_context_
, codec
, NULL
)) < 0) {
82 DLOG(WARNING
) << "AudioFileReader::Open() : could not open codec -"
83 << " result: " << result
;
87 // Ensure avcodec_open2() respected our format request.
88 if (codec_context_
->sample_fmt
== AV_SAMPLE_FMT_S16P
) {
89 DLOG(ERROR
) << "AudioFileReader::Open() : unable to configure a"
90 << " supported sample format - "
91 << codec_context_
->sample_fmt
;
95 DLOG(WARNING
) << "AudioFileReader::Open() : could not find codec -"
96 << " result: " << result
;
100 // Verify the channel layout is supported by Chrome. Acts as a sanity check
101 // against invalid files. See http://crbug.com/171962
102 if (ChannelLayoutToChromeChannelLayout(
103 codec_context_
->channel_layout
, codec_context_
->channels
) ==
104 CHANNEL_LAYOUT_UNSUPPORTED
) {
108 // Store initial values to guard against midstream configuration changes.
109 channels_
= codec_context_
->channels
;
110 sample_rate_
= codec_context_
->sample_rate
;
111 av_sample_format_
= codec_context_
->sample_fmt
;
116 void AudioFileReader::Close() {
117 if (codec_context_
) {
118 avcodec_close(codec_context_
);
119 codec_context_
= NULL
;
123 int AudioFileReader::Read(AudioBus
* audio_bus
) {
124 DCHECK(glue_
.get() && codec_context_
) <<
125 "AudioFileReader::Read() : reader is not opened!";
127 DCHECK_EQ(audio_bus
->channels(), channels());
128 if (audio_bus
->channels() != channels())
131 size_t bytes_per_sample
= av_get_bytes_per_sample(codec_context_
->sample_fmt
);
133 // Holds decoded audio.
134 scoped_ptr_malloc
<AVFrame
, ScopedPtrAVFree
> av_frame(avcodec_alloc_frame());
136 // Read until we hit EOF or we've read the requested number of frames.
138 int current_frame
= 0;
139 bool continue_decoding
= true;
141 while (current_frame
< audio_bus
->frames() && continue_decoding
&&
142 av_read_frame(glue_
->format_context(), &packet
) >= 0 &&
143 av_dup_packet(&packet
) >= 0) {
144 // Skip packets from other streams.
145 if (packet
.stream_index
!= stream_index_
) {
146 av_free_packet(&packet
);
150 // Make a shallow copy of packet so we can slide packet.data as frames are
151 // decoded from the packet; otherwise av_free_packet() will corrupt memory.
152 AVPacket packet_temp
= packet
;
154 avcodec_get_frame_defaults(av_frame
.get());
155 int frame_decoded
= 0;
156 int result
= avcodec_decode_audio4(
157 codec_context_
, av_frame
.get(), &frame_decoded
, &packet_temp
);
161 << "AudioFileReader::Read() : error in avcodec_decode_audio4() -"
163 continue_decoding
= false;
167 // Update packet size and data pointer in case we need to call the decoder
168 // with the remaining bytes from this packet.
169 packet_temp
.size
-= result
;
170 packet_temp
.data
+= result
;
175 // Determine the number of sample-frames we just decoded. Check overflow.
176 int frames_read
= av_frame
->nb_samples
;
177 if (frames_read
< 0) {
178 continue_decoding
= false;
182 #ifdef CHROMIUM_NO_AVFRAME_CHANNELS
183 int channels
= av_get_channel_layout_nb_channels(
184 av_frame
->channel_layout
);
186 int channels
= av_frame
->channels
;
188 if (av_frame
->sample_rate
!= sample_rate_
||
189 channels
!= channels_
||
190 av_frame
->format
!= av_sample_format_
) {
191 DLOG(ERROR
) << "Unsupported midstream configuration change!"
192 << " Sample Rate: " << av_frame
->sample_rate
<< " vs "
194 << ", Channels: " << channels
<< " vs "
196 << ", Sample Format: " << av_frame
->format
<< " vs "
197 << av_sample_format_
;
199 // This is an unrecoverable error, so bail out.
200 continue_decoding
= false;
204 // Truncate, if necessary, if the destination isn't big enough.
205 if (current_frame
+ frames_read
> audio_bus
->frames())
206 frames_read
= audio_bus
->frames() - current_frame
;
208 // Deinterleave each channel and convert to 32bit floating-point with
209 // nominal range -1.0 -> +1.0. If the output is already in float planar
210 // format, just copy it into the AudioBus.
211 if (codec_context_
->sample_fmt
== AV_SAMPLE_FMT_FLT
) {
212 float* decoded_audio_data
= reinterpret_cast<float*>(av_frame
->data
[0]);
213 int channels
= audio_bus
->channels();
214 for (int ch
= 0; ch
< channels
; ++ch
) {
215 float* bus_data
= audio_bus
->channel(ch
) + current_frame
;
216 for (int i
= 0, offset
= ch
; i
< frames_read
;
217 ++i
, offset
+= channels
) {
218 bus_data
[i
] = decoded_audio_data
[offset
];
221 } else if (codec_context_
->sample_fmt
== AV_SAMPLE_FMT_FLTP
) {
222 for (int ch
= 0; ch
< audio_bus
->channels(); ++ch
) {
223 memcpy(audio_bus
->channel(ch
) + current_frame
,
224 av_frame
->extended_data
[ch
], sizeof(float) * frames_read
);
227 audio_bus
->FromInterleavedPartial(
228 av_frame
->data
[0], current_frame
, frames_read
, bytes_per_sample
);
231 current_frame
+= frames_read
;
232 } while (packet_temp
.size
> 0);
233 av_free_packet(&packet
);
236 // Zero any remaining frames.
237 audio_bus
->ZeroFramesPartial(
238 current_frame
, audio_bus
->frames() - current_frame
);
240 // Returns the actual number of sample-frames decoded.
241 // Ideally this represents the "true" exact length of the file.
242 return current_frame
;