media/base/audio_buffer.cc

   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
   3 // found in the LICENSE file.
   4
   5 #include "media/base/audio_buffer.h"
   6
   7 #include "base/logging.h"
   8 #include "media/base/audio_bus.h"
   9 #include "media/base/buffers.h"
  10 #include "media/base/limits.h"
  11
  12 namespace media {
  13
  14 AudioBuffer::AudioBuffer(SampleFormat sample_format,
  15                          int channel_count,
  16                          int frame_count,
  17                          bool create_buffer,
  18                          const uint8* const* data,
  19                          const base::TimeDelta timestamp,
  20                          const base::TimeDelta duration)
  21     : sample_format_(sample_format),
  22       channel_count_(channel_count),
  23       adjusted_frame_count_(frame_count),
  24       trim_start_(0),
  25       end_of_stream_(!create_buffer && data == NULL && frame_count == 0),
  26       timestamp_(timestamp),
  27       duration_(duration) {
  28   CHECK_GE(channel_count, 0);
  29   CHECK_LE(channel_count, limits::kMaxChannels);
  30   CHECK_GE(frame_count, 0);
  31   int bytes_per_channel = SampleFormatToBytesPerChannel(sample_format);
  32   DCHECK_LE(bytes_per_channel, kChannelAlignment);
  33   int data_size = frame_count * bytes_per_channel;
  34
  35   // Empty buffer?
  36   if (!create_buffer)
  37     return;
  38
  39   if (sample_format == kSampleFormatPlanarF32 ||
  40       sample_format == kSampleFormatPlanarS16) {
  41     // Planar data, so need to allocate buffer for each channel.
  42     // Determine per channel data size, taking into account alignment.
  43     int block_size_per_channel =
  44         (data_size + kChannelAlignment - 1) & ~(kChannelAlignment - 1);
  45     DCHECK_GE(block_size_per_channel, data_size);
  46
  47     // Allocate a contiguous buffer for all the channel data.
  48     data_.reset(static_cast<uint8*>(base::AlignedAlloc(
  49         channel_count * block_size_per_channel, kChannelAlignment)));
  50     channel_data_.reserve(channel_count);
  51
  52     // Copy each channel's data into the appropriate spot.
  53     for (int i = 0; i < channel_count; ++i) {
  54       channel_data_.push_back(data_.get() + i * block_size_per_channel);
  55       if (data)
  56         memcpy(channel_data_[i], data[i], data_size);
  57     }
  58     return;
  59   }
  60
  61   // Remaining formats are interleaved data.
  62   DCHECK(sample_format_ == kSampleFormatU8 ||
  63          sample_format_ == kSampleFormatS16 ||
  64          sample_format_ == kSampleFormatS32 ||
  65          sample_format_ == kSampleFormatF32) << sample_format_;
  66   // Allocate our own buffer and copy the supplied data into it. Buffer must
  67   // contain the data for all channels.
  68   data_size *= channel_count;
  69   data_.reset(
  70       static_cast<uint8*>(base::AlignedAlloc(data_size, kChannelAlignment)));
  71   channel_data_.reserve(1);
  72   channel_data_.push_back(data_.get());
  73   if (data)
  74     memcpy(data_.get(), data[0], data_size);
  75 }
  76
  77 AudioBuffer::~AudioBuffer() {}
  78
  79 // static
  80 scoped_refptr<AudioBuffer> AudioBuffer::CopyFrom(
  81     SampleFormat sample_format,
  82     int channel_count,
  83     int frame_count,
  84     const uint8* const* data,
  85     const base::TimeDelta timestamp,
  86     const base::TimeDelta duration) {
  87   // If you hit this CHECK you likely have a bug in a demuxer. Go fix it.
  88   CHECK_GT(frame_count, 0);  // Otherwise looks like an EOF buffer.
  89   CHECK(data[0]);
  90   return make_scoped_refptr(new AudioBuffer(sample_format,
  91                                             channel_count,
  92                                             frame_count,
  93                                             true,
  94                                             data,
  95                                             timestamp,
  96                                             duration));
  97 }
  98
  99 // static
 100 scoped_refptr<AudioBuffer> AudioBuffer::CreateBuffer(SampleFormat sample_format,
 101                                                      int channel_count,
 102                                                      int frame_count) {
 103   CHECK_GT(frame_count, 0);  // Otherwise looks like an EOF buffer.
 104   return make_scoped_refptr(new AudioBuffer(sample_format,
 105                                             channel_count,
 106                                             frame_count,
 107                                             true,
 108                                             NULL,
 109                                             kNoTimestamp(),
 110                                             kNoTimestamp()));
 111 }
 112
 113 // static
 114 scoped_refptr<AudioBuffer> AudioBuffer::CreateEmptyBuffer(
 115     int channel_count,
 116     int frame_count,
 117     const base::TimeDelta timestamp,
 118     const base::TimeDelta duration) {
 119   CHECK_GT(frame_count, 0);  // Otherwise looks like an EOF buffer.
 120   // Since data == NULL, format doesn't matter.
 121   return make_scoped_refptr(new AudioBuffer(kSampleFormatF32,
 122                                             channel_count,
 123                                             frame_count,
 124                                             false,
 125                                             NULL,
 126                                             timestamp,
 127                                             duration));
 128 }
 129
 130 // static
 131 scoped_refptr<AudioBuffer> AudioBuffer::CreateEOSBuffer() {
 132   return make_scoped_refptr(new AudioBuffer(
 133       kUnknownSampleFormat, 1, 0, false, NULL, kNoTimestamp(), kNoTimestamp()));
 134 }
 135
 136 // Convert int16 values in the range [kint16min, kint16max] to [-1.0, 1.0].
 137 static inline float ConvertS16ToFloat(int16 value) {
 138   return value * (value < 0 ? -1.0f / kint16min : 1.0f / kint16max);
 139 }
 140
 141 void AudioBuffer::ReadFrames(int frames_to_copy,
 142                              int source_frame_offset,
 143                              int dest_frame_offset,
 144                              AudioBus* dest) {
 145   // Deinterleave each channel (if necessary) and convert to 32bit
 146   // floating-point with nominal range -1.0 -> +1.0 (if necessary).
 147
 148   // |dest| must have the same number of channels, and the number of frames
 149   // specified must be in range.
 150   DCHECK(!end_of_stream());
 151   DCHECK_EQ(dest->channels(), channel_count_);
 152   DCHECK_LE(source_frame_offset + frames_to_copy, adjusted_frame_count_);
 153   DCHECK_LE(dest_frame_offset + frames_to_copy, dest->frames());
 154
 155   // Move the start past any frames that have been trimmed.
 156   source_frame_offset += trim_start_;
 157
 158   if (!data_) {
 159     // Special case for an empty buffer.
 160     dest->ZeroFramesPartial(dest_frame_offset, frames_to_copy);
 161     return;
 162   }
 163
 164   if (sample_format_ == kSampleFormatPlanarF32) {
 165     // Format is planar float32. Copy the data from each channel as a block.
 166     for (int ch = 0; ch < channel_count_; ++ch) {
 167       const float* source_data =
 168           reinterpret_cast<const float*>(channel_data_[ch]) +
 169           source_frame_offset;
 170       memcpy(dest->channel(ch) + dest_frame_offset,
 171              source_data,
 172              sizeof(float) * frames_to_copy);
 173     }
 174     return;
 175   }
 176
 177   if (sample_format_ == kSampleFormatPlanarS16) {
 178     // Format is planar signed16. Convert each value into float and insert into
 179     // output channel data.
 180     for (int ch = 0; ch < channel_count_; ++ch) {
 181       const int16* source_data =
 182           reinterpret_cast<const int16*>(channel_data_[ch]) +
 183           source_frame_offset;
 184       float* dest_data = dest->channel(ch) + dest_frame_offset;
 185       for (int i = 0; i < frames_to_copy; ++i) {
 186         dest_data[i] = ConvertS16ToFloat(source_data[i]);
 187       }
 188     }
 189     return;
 190   }
 191
 192   if (sample_format_ == kSampleFormatF32) {
 193     // Format is interleaved float32. Copy the data into each channel.
 194     const float* source_data = reinterpret_cast<const float*>(data_.get()) +
 195                                source_frame_offset * channel_count_;
 196     for (int ch = 0; ch < channel_count_; ++ch) {
 197       float* dest_data = dest->channel(ch) + dest_frame_offset;
 198       for (int i = 0, offset = ch; i < frames_to_copy;
 199            ++i, offset += channel_count_) {
 200         dest_data[i] = source_data[offset];
 201       }
 202     }
 203     return;
 204   }
 205
 206   // Remaining formats are integer interleaved data. Use the deinterleaving code
 207   // in AudioBus to copy the data.
 208   DCHECK(sample_format_ == kSampleFormatU8 ||
 209          sample_format_ == kSampleFormatS16 ||
 210          sample_format_ == kSampleFormatS32);
 211   int bytes_per_channel = SampleFormatToBytesPerChannel(sample_format_);
 212   int frame_size = channel_count_ * bytes_per_channel;
 213   const uint8* source_data = data_.get() + source_frame_offset * frame_size;
 214   dest->FromInterleavedPartial(
 215       source_data, dest_frame_offset, frames_to_copy, bytes_per_channel);
 216 }
 217
 218 void AudioBuffer::TrimStart(int frames_to_trim) {
 219   CHECK_GE(frames_to_trim, 0);
 220   CHECK_LE(frames_to_trim, adjusted_frame_count_);
 221
 222   // Adjust timestamp_ and duration_ to reflect the smaller number of frames.
 223   double offset = static_cast<double>(duration_.InMicroseconds()) *
 224                   frames_to_trim / adjusted_frame_count_;
 225   base::TimeDelta offset_as_time =
 226       base::TimeDelta::FromMicroseconds(static_cast<int64>(offset));
 227   timestamp_ += offset_as_time;
 228   duration_ -= offset_as_time;
 229
 230   // Finally adjust the number of frames in this buffer and where the start
 231   // really is.
 232   adjusted_frame_count_ -= frames_to_trim;
 233   trim_start_ += frames_to_trim;
 234 }
 235
 236 void AudioBuffer::TrimEnd(int frames_to_trim) {
 237   CHECK_GE(frames_to_trim, 0);
 238   CHECK_LE(frames_to_trim, adjusted_frame_count_);
 239
 240   // Adjust duration_ only to reflect the smaller number of frames.
 241   double offset = static_cast<double>(duration_.InMicroseconds()) *
 242                   frames_to_trim / adjusted_frame_count_;
 243   base::TimeDelta offset_as_time =
 244       base::TimeDelta::FromMicroseconds(static_cast<int64>(offset));
 245   duration_ -= offset_as_time;
 246
 247   // Finally adjust the number of frames in this buffer.
 248   adjusted_frame_count_ -= frames_to_trim;
 249 }
 250
 251 }  // namespace media