Fix muted audio when playback rate != 1.0 or 0.0

media/audio/audio_util.cc

 // 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.
 
 // Software adjust volume of samples, allows each audio stream its own
 // volume without impacting master volume for chrome and other applications.
 
 // Implemented as templates to allow 8, 16 and 32 bit implementations.
 // 8 bit is unsigned and biased by 128.
 
@@ skipping 33 matching lines @@
 template<class Format, class Fixed, int bias>
 static void AdjustVolume(Format* buf_out,
                          int sample_count,
                          int fixed_volume) {
   for (int i = 0; i < sample_count; ++i) {
     buf_out[i] = static_cast<Format>(ScaleChannel<Fixed>(buf_out[i] - bias,
                                                          fixed_volume) + bias);
   }
 }
 
-// Type is the datatype of a data point in the waveform (i.e. uint8, int16,
-// int32, etc).
-template <class Type>
-static void DoCrossfade(int bytes_to_crossfade, int number_of_channels,
-                        int bytes_per_channel, const Type* src, Type* dest) {
-  DCHECK_EQ(sizeof(Type), static_cast<size_t>(bytes_per_channel));
-  int number_of_samples =
-      bytes_to_crossfade / (bytes_per_channel * number_of_channels);
-
-  const Type* dest_end = dest + number_of_samples * number_of_channels;
-  const Type* src_end = src + number_of_samples * number_of_channels;
-
-  for (int i = 0; i < number_of_samples; ++i) {
-    double crossfade_ratio = static_cast<double>(i) / number_of_samples;
-    for (int j = 0; j < number_of_channels; ++j) {
-      DCHECK_LT(dest, dest_end);
-      DCHECK_LT(src, src_end);
-      *dest = (*dest) * (1.0 - crossfade_ratio) + (*src) * crossfade_ratio;
-      ++src;
-      ++dest;
-    }
-  }
-}
-
 static const int kChannel_L = 0;
 static const int kChannel_R = 1;
 static const int kChannel_C = 2;
 
 template<class Fixed, int min_value, int max_value>
 static int AddChannel(int val, int adder) {
   Fixed sum = static_cast<Fixed>(val) + static_cast<Fixed>(adder);
   if (sum > max_value)
     return max_value;
   if (sum < min_value)
@@ skipping 320 matching lines @@
 #if defined(OS_WIN)
 
 bool IsWASAPISupported() {
   // Note: that function correctly returns that Windows Server 2003 does not
   // support WASAPI.
   return base::win::GetVersion() >= base::win::VERSION_VISTA;
 }
 
 #endif
 
-void Crossfade(int bytes_to_crossfade, int number_of_channels,
-               int bytes_per_channel, const uint8* src, uint8* dest) {
-  // TODO(vrk): The type punning below is no good!
-  switch (bytes_per_channel) {
-    case 4:
-      DoCrossfade(bytes_to_crossfade, number_of_channels, bytes_per_channel,
-                  reinterpret_cast<const int32*>(src),
-                  reinterpret_cast<int32*>(dest));
-      break;
-    case 2:
-      DoCrossfade(bytes_to_crossfade, number_of_channels, bytes_per_channel,
-                  reinterpret_cast<const int16*>(src),
-                  reinterpret_cast<int16*>(dest));
-      break;
-    case 1:
-      DoCrossfade(bytes_to_crossfade, number_of_channels, bytes_per_channel,
-                  src, dest);
-      break;
-    default:
-      NOTREACHED() << "Unsupported audio bit depth in crossfade.";
-  }
-}
-
-// The minimum number of samples in a hardware packet.
-// This value is selected so that we can handle down to 5khz sample rate.
-static const int kMinSamplesPerHardwarePacket = 1024;
-
-// The maximum number of samples in a hardware packet.
-// This value is selected so that we can handle up to 192khz sample rate.
-static const int kMaxSamplesPerHardwarePacket = 64 * 1024;
-
-// This constant governs the hardware audio buffer size, this value should be
-// chosen carefully.
-// This value is selected so that we have 8192 samples for 48khz streams.
-static const int kMillisecondsPerHardwarePacket = 170;
-
-uint32 SelectSamplesPerPacket(int sample_rate) {
-  // Select the number of samples that can provide at least
-  // |kMillisecondsPerHardwarePacket| worth of audio data.
-  int samples = kMinSamplesPerHardwarePacket;
-  while (samples <= kMaxSamplesPerHardwarePacket &&
-         samples * base::Time::kMillisecondsPerSecond <
-         sample_rate * kMillisecondsPerHardwarePacket) {
-    samples *= 2;
-  }
-  return samples;
-}
-
 }  // namespace media