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
|
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#include "AudioSegment.h"
#include "AudioMixer.h"
#include "AudioChannelFormat.h"
#include "Latency.h"
#include <speex/speex_resampler.h>
namespace mozilla {
const uint8_t SilentChannel::gZeroChannel[MAX_AUDIO_SAMPLE_SIZE*SilentChannel::AUDIO_PROCESSING_FRAMES] = {0};
template<>
const float* SilentChannel::ZeroChannel<float>()
{
return reinterpret_cast<const float*>(SilentChannel::gZeroChannel);
}
template<>
const int16_t* SilentChannel::ZeroChannel<int16_t>()
{
return reinterpret_cast<const int16_t*>(SilentChannel::gZeroChannel);
}
void
AudioSegment::ApplyVolume(float aVolume)
{
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
ci->mVolume *= aVolume;
}
}
void AudioSegment::ResampleChunks(SpeexResamplerState* aResampler, uint32_t aInRate, uint32_t aOutRate)
{
if (mChunks.IsEmpty()) {
return;
}
MOZ_ASSERT(aResampler || IsNull(), "We can only be here without a resampler if this segment is null.");
AudioSampleFormat format = AUDIO_FORMAT_SILENCE;
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
if (ci->mBufferFormat != AUDIO_FORMAT_SILENCE) {
format = ci->mBufferFormat;
}
}
switch (format) {
// If the format is silence at this point, all the chunks are silent. The
// actual function we use does not matter, it's just a matter of changing
// the chunks duration.
case AUDIO_FORMAT_SILENCE:
case AUDIO_FORMAT_FLOAT32:
Resample<float>(aResampler, aInRate, aOutRate);
break;
case AUDIO_FORMAT_S16:
Resample<int16_t>(aResampler, aInRate, aOutRate);
break;
default:
MOZ_ASSERT(false);
break;
}
}
// This helps to to safely get a pointer to the position we want to start
// writing a planar audio buffer, depending on the channel and the offset in the
// buffer.
static AudioDataValue*
PointerForOffsetInChannel(AudioDataValue* aData, size_t aLengthSamples,
uint32_t aChannelCount, uint32_t aChannel,
uint32_t aOffsetSamples)
{
size_t samplesPerChannel = aLengthSamples / aChannelCount;
size_t beginningOfChannel = samplesPerChannel * aChannel;
MOZ_ASSERT(aChannel * samplesPerChannel + aOffsetSamples < aLengthSamples,
"Offset request out of bounds.");
return aData + beginningOfChannel + aOffsetSamples;
}
void
AudioSegment::Mix(AudioMixer& aMixer, uint32_t aOutputChannels,
uint32_t aSampleRate)
{
AutoTArray<AudioDataValue, SilentChannel::AUDIO_PROCESSING_FRAMES* GUESS_AUDIO_CHANNELS>
buf;
AutoTArray<const AudioDataValue*, GUESS_AUDIO_CHANNELS> channelData;
uint32_t offsetSamples = 0;
uint32_t duration = GetDuration();
if (duration <= 0) {
MOZ_ASSERT(duration == 0);
return;
}
uint32_t outBufferLength = duration * aOutputChannels;
buf.SetLength(outBufferLength);
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
AudioChunk& c = *ci;
uint32_t frames = c.mDuration;
// If the chunk is silent, simply write the right number of silence in the
// buffers.
if (c.mBufferFormat == AUDIO_FORMAT_SILENCE) {
for (uint32_t channel = 0; channel < aOutputChannels; channel++) {
AudioDataValue* ptr =
PointerForOffsetInChannel(buf.Elements(), outBufferLength,
aOutputChannels, channel, offsetSamples);
PodZero(ptr, frames);
}
} else {
// Othewise, we need to upmix or downmix appropriately, depending on the
// desired input and output channels.
channelData.SetLength(c.mChannelData.Length());
for (uint32_t i = 0; i < channelData.Length(); ++i) {
channelData[i] = static_cast<const AudioDataValue*>(c.mChannelData[i]);
}
if (channelData.Length() < aOutputChannels) {
// Up-mix.
AudioChannelsUpMix(&channelData, aOutputChannels, SilentChannel::ZeroChannel<AudioDataValue>());
for (uint32_t channel = 0; channel < aOutputChannels; channel++) {
AudioDataValue* ptr =
PointerForOffsetInChannel(buf.Elements(), outBufferLength,
aOutputChannels, channel, offsetSamples);
PodCopy(ptr, reinterpret_cast<const AudioDataValue*>(channelData[channel]),
frames);
}
MOZ_ASSERT(channelData.Length() == aOutputChannels);
} else if (channelData.Length() > aOutputChannels) {
// Down mix.
AutoTArray<AudioDataValue*, GUESS_AUDIO_CHANNELS> outChannelPtrs;
outChannelPtrs.SetLength(aOutputChannels);
uint32_t offsetSamples = 0;
for (uint32_t channel = 0; channel < aOutputChannels; channel++) {
outChannelPtrs[channel] =
PointerForOffsetInChannel(buf.Elements(), outBufferLength,
aOutputChannels, channel, offsetSamples);
}
AudioChannelsDownMix(channelData, outChannelPtrs.Elements(),
aOutputChannels, frames);
} else {
// The channel count is already what we want, just copy it over.
for (uint32_t channel = 0; channel < aOutputChannels; channel++) {
AudioDataValue* ptr =
PointerForOffsetInChannel(buf.Elements(), outBufferLength,
aOutputChannels, channel, offsetSamples);
PodCopy(ptr, reinterpret_cast<const AudioDataValue*>(channelData[channel]),
frames);
}
}
}
offsetSamples += frames;
}
if (offsetSamples) {
MOZ_ASSERT(offsetSamples == outBufferLength / aOutputChannels,
"We forgot to write some samples?");
aMixer.Mix(buf.Elements(), aOutputChannels, offsetSamples, aSampleRate);
}
}
void
AudioSegment::WriteTo(uint64_t aID, AudioMixer& aMixer, uint32_t aOutputChannels, uint32_t aSampleRate)
{
AutoTArray<AudioDataValue,SilentChannel::AUDIO_PROCESSING_FRAMES*GUESS_AUDIO_CHANNELS> buf;
// Offset in the buffer that will be written to the mixer, in samples.
uint32_t offset = 0;
if (GetDuration() <= 0) {
MOZ_ASSERT(GetDuration() == 0);
return;
}
uint32_t outBufferLength = GetDuration() * aOutputChannels;
buf.SetLength(outBufferLength);
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
AudioChunk& c = *ci;
switch (c.mBufferFormat) {
case AUDIO_FORMAT_S16:
WriteChunk<int16_t>(c, aOutputChannels, buf.Elements() + offset);
break;
case AUDIO_FORMAT_FLOAT32:
WriteChunk<float>(c, aOutputChannels, buf.Elements() + offset);
break;
case AUDIO_FORMAT_SILENCE:
// The mixer is expecting interleaved data, so this is ok.
PodZero(buf.Elements() + offset, c.mDuration * aOutputChannels);
break;
default:
MOZ_ASSERT(false, "Not handled");
}
offset += c.mDuration * aOutputChannels;
if (!c.mTimeStamp.IsNull()) {
TimeStamp now = TimeStamp::Now();
// would be more efficient to c.mTimeStamp to ms on create time then pass here
LogTime(AsyncLatencyLogger::AudioMediaStreamTrack, aID,
(now - c.mTimeStamp).ToMilliseconds(), c.mTimeStamp);
}
}
if (offset) {
aMixer.Mix(buf.Elements(), aOutputChannels, offset / aOutputChannels, aSampleRate);
}
}
} // namespace mozilla
|
