summaryrefslogtreecommitdiffstats
path: root/dom/media/AudioSegment.cpp
blob: 243cdffd0e09ea9ce5eac3e8ba95a4cc6a900b12 (plain)
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
/* -*- 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 "MediaTrackGraph.h"  // for nsAutoRefTraits<SpeexResamplerState>
#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;
  }
}

template <typename T>
void AudioSegment::Resample(nsAutoRef<SpeexResamplerState>& aResampler,
                            uint32_t* aResamplerChannelCount, uint32_t aInRate,
                            uint32_t aOutRate) {
  mDuration = 0;

  for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
    AutoTArray<nsTArray<T>, GUESS_AUDIO_CHANNELS> output;
    AutoTArray<const T*, GUESS_AUDIO_CHANNELS> bufferPtrs;
    AudioChunk& c = *ci;
    // If this chunk is null, don't bother resampling, just alter its duration
    if (c.IsNull()) {
      c.mDuration = (c.mDuration * aOutRate) / aInRate;
      mDuration += c.mDuration;
      continue;
    }
    uint32_t channels = c.mChannelData.Length();
    // This might introduce a discontinuity, but a channel count change in the
    // middle of a stream is not that common. This also initializes the
    // resampler as late as possible.
    if (channels != *aResamplerChannelCount) {
      SpeexResamplerState* state =
          speex_resampler_init(channels, aInRate, aOutRate,
                               SPEEX_RESAMPLER_QUALITY_DEFAULT, nullptr);
      MOZ_ASSERT(state);
      aResampler.own(state);
      *aResamplerChannelCount = channels;
    }
    output.SetLength(channels);
    bufferPtrs.SetLength(channels);
    uint32_t inFrames = c.mDuration;
    // Round up to allocate; the last frame may not be used.
    NS_ASSERTION((UINT64_MAX - aInRate + 1) / c.mDuration >= aOutRate,
                 "Dropping samples");
    uint32_t outSize =
        (static_cast<uint64_t>(c.mDuration) * aOutRate + aInRate - 1) / aInRate;
    for (uint32_t i = 0; i < channels; i++) {
      T* out = output[i].AppendElements(outSize);
      uint32_t outFrames = outSize;

      const T* in = static_cast<const T*>(c.mChannelData[i]);
      dom::WebAudioUtils::SpeexResamplerProcess(aResampler.get(), i, in,
                                                &inFrames, out, &outFrames);
      MOZ_ASSERT(inFrames == c.mDuration);

      bufferPtrs[i] = out;
      output[i].SetLength(outFrames);
    }
    MOZ_ASSERT(channels > 0);
    c.mDuration = output[0].Length();
    c.mBuffer = new mozilla::SharedChannelArrayBuffer<T>(std::move(output));
    for (uint32_t i = 0; i < channels; i++) {
      c.mChannelData[i] = bufferPtrs[i];
    }
    mDuration += c.mDuration;
  }
}

void AudioSegment::ResampleChunks(nsAutoRef<SpeexResamplerState>& aResampler,
                                  uint32_t* aResamplerChannelCount,
                                  uint32_t aInRate, uint32_t aOutRate) {
  if (mChunks.IsEmpty()) {
    return;
  }

  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, aResamplerChannelCount, aInRate, aOutRate);
      break;
    case AUDIO_FORMAT_S16:
      Resample<int16_t>(aResampler, aResamplerChannelCount, aInRate, aOutRate);
      break;
    default:
      MOZ_ASSERT(false);
      break;
  }
}

size_t AudioSegment::WriteToInterleavedBuffer(nsTArray<AudioDataValue>& aBuffer,
                                              uint32_t aChannels) const {
  size_t offset = 0;
  if (GetDuration() <= 0) {
    MOZ_ASSERT(GetDuration() == 0);
    return offset;
  }

  // Calculate how many samples in this segment
  size_t frames = static_cast<size_t>(GetDuration());
  CheckedInt<size_t> samples(frames);
  samples *= static_cast<size_t>(aChannels);
  MOZ_ASSERT(samples.isValid());
  if (!samples.isValid()) {
    return offset;
  }

  // Enlarge buffer space if needed
  if (samples.value() > aBuffer.Capacity()) {
    aBuffer.SetCapacity(samples.value());
  }
  aBuffer.SetLengthAndRetainStorage(samples.value());
  aBuffer.ClearAndRetainStorage();

  // Convert the de-interleaved chunks into an interleaved buffer. Note that
  // we may upmix or downmix the audio data if the channel in the chunks
  // mismatch with aChannels
  for (ConstChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
    const AudioChunk& c = *ci;
    size_t samplesInChunk = static_cast<size_t>(c.mDuration) * aChannels;
    switch (c.mBufferFormat) {
      case AUDIO_FORMAT_S16:
        WriteChunk<int16_t>(c, aChannels, c.mVolume,
                            aBuffer.Elements() + offset);
        break;
      case AUDIO_FORMAT_FLOAT32:
        WriteChunk<float>(c, aChannels, c.mVolume, aBuffer.Elements() + offset);
        break;
      case AUDIO_FORMAT_SILENCE:
        PodZero(aBuffer.Elements() + offset, samplesInChunk);
        break;
      default:
        MOZ_ASSERT_UNREACHABLE("Unknown format");
        PodZero(aBuffer.Elements() + offset, samplesInChunk);
        break;
    }
    offset += samplesInChunk;
  }
  MOZ_DIAGNOSTIC_ASSERT(samples.value() == offset,
                        "Segment's duration is incorrect");
  aBuffer.SetLengthAndRetainStorage(offset);
  return offset;
}

// 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;
}

template <typename SrcT>
static void DownMixChunk(const AudioChunk& aChunk,
                         Span<AudioDataValue* const> aOutputChannels) {
  Span<const SrcT* const> channelData = aChunk.ChannelData<SrcT>();
  uint32_t frameCount = aChunk.mDuration;
  if (channelData.Length() > aOutputChannels.Length()) {
    // Down mix.
    AudioChannelsDownMix(channelData, aOutputChannels, frameCount);
    for (AudioDataValue* outChannel : aOutputChannels) {
      ScaleAudioSamples(outChannel, frameCount, aChunk.mVolume);
    }
  } else {
    // The channel count is already what we want.
    for (uint32_t channel = 0; channel < aOutputChannels.Length(); channel++) {
      ConvertAudioSamplesWithScale(channelData[channel],
                                   aOutputChannels[channel], frameCount,
                                   aChunk.mVolume);
    }
  }
}

void AudioChunk::DownMixTo(
    Span<AudioDataValue* const> aOutputChannelPtrs) const {
  switch (mBufferFormat) {
    case AUDIO_FORMAT_FLOAT32:
      DownMixChunk<float>(*this, aOutputChannelPtrs);
      return;
    case AUDIO_FORMAT_S16:
      DownMixChunk<int16_t>(*this, aOutputChannelPtrs);
      return;
    case AUDIO_FORMAT_SILENCE:
      for (AudioDataValue* outChannel : aOutputChannelPtrs) {
        std::fill_n(outChannel, mDuration, static_cast<AudioDataValue>(0));
      }
      return;
      // Avoid `default:` so that `-Wswitch` catches missing enumerators at
      // compile time.
  }
  MOZ_ASSERT_UNREACHABLE("buffer format");
}

void AudioSegment::Mix(AudioMixer& aMixer, uint32_t aOutputChannels,
                       uint32_t aSampleRate) {
  AutoTArray<AudioDataValue,
             SilentChannel::AUDIO_PROCESSING_FRAMES * GUESS_AUDIO_CHANNELS>
      buf;
  AudioChunk upMixChunk;
  uint32_t offsetSamples = 0;
  uint32_t duration = GetDuration();

  if (duration <= 0) {
    MOZ_ASSERT(duration == 0);
    return;
  }

  uint32_t outBufferLength = duration * aOutputChannels;
  buf.SetLength(outBufferLength);

  AutoTArray<AudioDataValue*, GUESS_AUDIO_CHANNELS> outChannelPtrs;
  outChannelPtrs.SetLength(aOutputChannels);

  uint32_t frames;
  for (ChunkIterator ci(*this); !ci.IsEnded();
       ci.Next(), offsetSamples += frames) {
    const AudioChunk& c = *ci;
    frames = c.mDuration;
    for (uint32_t channel = 0; channel < aOutputChannels; channel++) {
      outChannelPtrs[channel] =
          PointerForOffsetInChannel(buf.Elements(), outBufferLength,
                                    aOutputChannels, channel, offsetSamples);
    }

    // If the chunk is silent, simply write the right number of silence in the
    // buffers.
    if (c.mBufferFormat == AUDIO_FORMAT_SILENCE) {
      for (AudioDataValue* outChannel : outChannelPtrs) {
        PodZero(outChannel, frames);
      }
      continue;
    }
    // We need to upmix and downmix appropriately, depending on the
    // desired input and output channels.
    const AudioChunk* downMixInput = &c;
    if (c.ChannelCount() < aOutputChannels) {
      // Up-mix.
      upMixChunk = c;
      AudioChannelsUpMix<void>(&upMixChunk.mChannelData, aOutputChannels,
                               SilentChannel::gZeroChannel);
      downMixInput = &upMixChunk;
    }
    downMixInput->DownMixTo(outChannelPtrs);
  }

  if (offsetSamples) {
    MOZ_ASSERT(offsetSamples == outBufferLength / aOutputChannels,
               "We forgot to write some samples?");
    aMixer.Mix(buf.Elements(), aOutputChannels, offsetSamples, aSampleRate);
  }
}

}  // namespace mozilla