Remove the last piece of deprecated synchronous IO code.

BUG=337096
TEST=bots

Review URL: https://codereview.chromium.org/460373002

Cr-Commit-Position: refs/heads/master@{#292132}

19 files changed, 27 insertions, 155 deletions

diff --git a/media/audio/alsa/audio_manager_alsa.cc b/media/audio/alsa/audio_manager_alsa.cc
index beb60ba..7624834 100644
--- a/media/audio/alsa/audio_manager_alsa.cc
+++ b/media/audio/alsa/audio_manager_alsa.cc
@@ -311,7 +311,6 @@ AudioParameters AudioManagerAlsa::GetPreferredOutputStreamParameters(
   int sample_rate = kDefaultSampleRate;
   int buffer_size = kDefaultOutputBufferSize;
   int bits_per_sample = 16;
-  int input_channels = 0;
   if (input_params.IsValid()) {
     // Some clients, such as WebRTC, have a more limited use case and work
     // acceptably with a smaller buffer size.  The check below allows clients
@@ -321,7 +320,6 @@ AudioParameters AudioManagerAlsa::GetPreferredOutputStreamParameters(
     sample_rate = input_params.sample_rate();
     bits_per_sample = input_params.bits_per_sample();
     channel_layout = input_params.channel_layout();
-    input_channels = input_params.input_channels();
     buffer_size = std::min(input_params.frames_per_buffer(), buffer_size);
   }
 
@@ -330,7 +328,7 @@ AudioParameters AudioManagerAlsa::GetPreferredOutputStreamParameters(
     buffer_size = user_buffer_size;
 
   return AudioParameters(
-      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, input_channels,
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
       sample_rate, bits_per_sample, buffer_size, AudioParameters::NO_EFFECTS);
 }
 
diff --git a/media/audio/android/audio_android_unittest.cc b/media/audio/android/audio_android_unittest.cc
index 177953d..68e74a4 100644
--- a/media/audio/android/audio_android_unittest.cc
+++ b/media/audio/android/audio_android_unittest.cc
@@ -598,7 +598,6 @@ class AudioAndroidInputTest : public AudioAndroidOutputTest,
                                AudioParameters::NO_EFFECTS;
     AudioParameters params(audio_input_parameters().format(),
                            audio_input_parameters().channel_layout(),
-                           audio_input_parameters().input_channels(),
                            audio_input_parameters().sample_rate(),
                            audio_input_parameters().bits_per_sample(),
                            audio_input_parameters().frames_per_buffer(),
@@ -805,7 +804,6 @@ TEST_P(AudioAndroidInputTest,
   AudioParameters native_params = GetInputStreamParameters();
   AudioParameters params(native_params.format(),
                          native_params.channel_layout(),
-                         native_params.input_channels(),
                          native_params.sample_rate(),
                          native_params.bits_per_sample(),
                          native_params.sample_rate() / 100,
diff --git a/media/audio/android/audio_manager_android.cc b/media/audio/android/audio_manager_android.cc
index 48f203a..090f853 100644
--- a/media/audio/android/audio_manager_android.cc
+++ b/media/audio/android/audio_manager_android.cc
@@ -138,7 +138,7 @@ AudioParameters AudioManagerAndroid::GetInputStreamParameters(
     buffer_size = user_buffer_size;
 
   AudioParameters params(
-      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, 0,
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
       GetNativeOutputSampleRate(), 16, buffer_size, effects);
   return params;
 }
@@ -274,13 +274,11 @@ AudioParameters AudioManagerAndroid::GetPreferredOutputStreamParameters(
   int sample_rate = GetNativeOutputSampleRate();
   int buffer_size = GetOptimalOutputFrameSize(sample_rate, 2);
   int bits_per_sample = 16;
-  int input_channels = 0;
   if (input_params.IsValid()) {
     // Use the client's input parameters if they are valid.
     sample_rate = input_params.sample_rate();
     bits_per_sample = input_params.bits_per_sample();
     channel_layout = input_params.channel_layout();
-    input_channels = input_params.input_channels();
     buffer_size = GetOptimalOutputFrameSize(
         sample_rate, ChannelLayoutToChannelCount(channel_layout));
   }
@@ -290,7 +288,7 @@ AudioParameters AudioManagerAndroid::GetPreferredOutputStreamParameters(
     buffer_size = user_buffer_size;
 
   return AudioParameters(
-      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, input_channels,
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
       sample_rate, bits_per_sample, buffer_size, AudioParameters::NO_EFFECTS);
 }
 
diff --git a/media/audio/audio_output_device.h b/media/audio/audio_output_device.h
index 8449e9f..0d53d8b 100644
--- a/media/audio/audio_output_device.h
+++ b/media/audio/audio_output_device.h
@@ -81,11 +81,8 @@ class MEDIA_EXPORT AudioOutputDevice
       scoped_ptr<AudioOutputIPC> ipc,
       const scoped_refptr<base::SingleThreadTaskRunner>& io_task_runner);
 
-  // Initialize function for clients wishing to have unified input and
-  // output, |params| may specify |input_channels| > 0, representing a
-  // number of input channels which will be at the same sample-rate
-  // and buffer-size as the output as specified in |params|. |session_id| is
-  // used for the browser to select the correct input device.
+  // Initialize the stream using |session_id|, which is used for the browser
+  // to select the correct input device.
   void InitializeWithSessionId(const AudioParameters& params,
                                RenderCallback* callback,
                                int session_id);
diff --git a/media/audio/audio_output_device_unittest.cc b/media/audio/audio_output_device_unittest.cc
index 92d9425..0e1fd1e5 100644
--- a/media/audio/audio_output_device_unittest.cc
+++ b/media/audio/audio_output_device_unittest.cc
@@ -126,7 +126,7 @@ int AudioOutputDeviceTest::CalculateMemorySize() {
 AudioOutputDeviceTest::AudioOutputDeviceTest() {
   default_audio_parameters_.Reset(
       AudioParameters::AUDIO_PCM_LINEAR,
-      CHANNEL_LAYOUT_STEREO, 2, 0, 48000, 16, 1024);
+      CHANNEL_LAYOUT_STEREO, 2, 48000, 16, 1024);
 
   audio_output_ipc_ = new MockAudioOutputIPC();
   audio_device_ = new AudioOutputDevice(
diff --git a/media/audio/audio_output_resampler.cc b/media/audio/audio_output_resampler.cc
index 15633bb..cf966e0 100644
--- a/media/audio/audio_output_resampler.cc
+++ b/media/audio/audio_output_resampler.cc
@@ -221,7 +221,7 @@ bool AudioOutputResampler::OpenStream() {
   // Finally fall back to a fake audio output device.
   output_params_.Reset(
       AudioParameters::AUDIO_FAKE, params_.channel_layout(),
-      params_.channels(), params_.input_channels(), params_.sample_rate(),
+      params_.channels(), params_.sample_rate(),
       params_.bits_per_sample(), params_.frames_per_buffer());
   Initialize();
   if (dispatcher_->OpenStream()) {
diff --git a/media/audio/audio_parameters.cc b/media/audio/audio_parameters.cc
index 62b825f..7532f34 100644
--- a/media/audio/audio_parameters.cc
+++ b/media/audio/audio_parameters.cc
@@ -16,7 +16,6 @@ AudioParameters::AudioParameters()
       bits_per_sample_(0),
       frames_per_buffer_(0),
       channels_(0),
-      input_channels_(0),
       effects_(NO_EFFECTS) {
 }
 
@@ -29,12 +28,10 @@ AudioParameters::AudioParameters(Format format, ChannelLayout channel_layout,
       bits_per_sample_(bits_per_sample),
       frames_per_buffer_(frames_per_buffer),
       channels_(ChannelLayoutToChannelCount(channel_layout)),
-      input_channels_(0),
       effects_(NO_EFFECTS) {
 }
 
 AudioParameters::AudioParameters(Format format, ChannelLayout channel_layout,
-                                 int input_channels,
                                  int sample_rate, int bits_per_sample,
                                  int frames_per_buffer, int effects)
     : format_(format),
@@ -43,37 +40,33 @@ AudioParameters::AudioParameters(Format format, ChannelLayout channel_layout,
       bits_per_sample_(bits_per_sample),
       frames_per_buffer_(frames_per_buffer),
       channels_(ChannelLayoutToChannelCount(channel_layout)),
-      input_channels_(input_channels),
       effects_(effects) {
 }
 
 AudioParameters::AudioParameters(Format format, ChannelLayout channel_layout,
-                                 int channels, int input_channels,
-                                 int sample_rate, int bits_per_sample,
-                                 int frames_per_buffer, int effects)
+                                 int channels, int sample_rate,
+                                 int bits_per_sample, int frames_per_buffer,
+                                 int effects)
     : format_(format),
       channel_layout_(channel_layout),
       sample_rate_(sample_rate),
       bits_per_sample_(bits_per_sample),
       frames_per_buffer_(frames_per_buffer),
       channels_(channels),
-      input_channels_(input_channels),
       effects_(effects) {
   if (channel_layout != CHANNEL_LAYOUT_DISCRETE)
     DCHECK_EQ(channels, ChannelLayoutToChannelCount(channel_layout));
 }
 
 void AudioParameters::Reset(Format format, ChannelLayout channel_layout,
-                            int channels, int input_channels,
-                            int sample_rate, int bits_per_sample,
-                            int frames_per_buffer) {
+                            int channels, int sample_rate,
+                            int bits_per_sample, int frames_per_buffer) {
   if (channel_layout != CHANNEL_LAYOUT_DISCRETE)
     DCHECK_EQ(channels, ChannelLayoutToChannelCount(channel_layout));
 
   format_ = format;
   channel_layout_ = channel_layout;
   channels_ = channels;
-  input_channels_ = input_channels;
   sample_rate_ = sample_rate;
   bits_per_sample_ = bits_per_sample;
   frames_per_buffer_ = frames_per_buffer;
@@ -86,8 +79,6 @@ bool AudioParameters::IsValid() const {
          (channels_ <= media::limits::kMaxChannels) &&
          (channel_layout_ > CHANNEL_LAYOUT_UNSUPPORTED) &&
          (channel_layout_ <= CHANNEL_LAYOUT_MAX) &&
-         (input_channels_ >= 0) &&
-         (input_channels_ <= media::limits::kMaxChannels) &&
          (sample_rate_ >= media::limits::kMinSampleRate) &&
          (sample_rate_ <= media::limits::kMaxSampleRate) &&
          (bits_per_sample_ > 0) &&
diff --git a/media/audio/audio_parameters.h b/media/audio/audio_parameters.h
index b23d26f..03fe043 100644
--- a/media/audio/audio_parameters.h
+++ b/media/audio/audio_parameters.h
@@ -58,17 +58,14 @@ class MEDIA_EXPORT AudioParameters {
                   int sample_rate, int bits_per_sample,
                   int frames_per_buffer);
   AudioParameters(Format format, ChannelLayout channel_layout,
-                  int input_channels,
                   int sample_rate, int bits_per_sample,
                   int frames_per_buffer, int effects);
   AudioParameters(Format format, ChannelLayout channel_layout,
-                  int channels, int input_channels,
-                  int sample_rate, int bits_per_sample,
+                  int channels, int sample_rate, int bits_per_sample,
                   int frames_per_buffer, int effects);
 
   void Reset(Format format, ChannelLayout channel_layout,
-             int channels, int input_channels,
-             int sample_rate, int bits_per_sample,
+             int channels, int sample_rate, int bits_per_sample,
              int frames_per_buffer);
 
   // Checks that all values are in the expected range. All limits are specified
@@ -94,7 +91,6 @@ class MEDIA_EXPORT AudioParameters {
   int bits_per_sample() const { return bits_per_sample_; }
   int frames_per_buffer() const { return frames_per_buffer_; }
   int channels() const { return channels_; }
-  int input_channels() const { return input_channels_; }
   int effects() const { return effects_; }
 
   // Comparison with other AudioParams.
@@ -103,7 +99,6 @@ class MEDIA_EXPORT AudioParameters {
            sample_rate_ == other.sample_rate() &&
            channel_layout_ == other.channel_layout() &&
            channels_ == other.channels() &&
-           input_channels_ == other.input_channels() &&
            bits_per_sample_ == other.bits_per_sample() &&
            frames_per_buffer_ == other.frames_per_buffer() &&
            effects_ == other.effects();
@@ -120,9 +115,6 @@ class MEDIA_EXPORT AudioParameters {
 
   int channels_;                  // Number of channels. Value set based on
                                   // |channel_layout|.
-  int input_channels_;            // Optional number of input channels.
-                                  // Normally 0, but can be set to specify
-                                  // synchronized I/O.
   int effects_;                   // Bitmask using PlatformEffectsMask.
 };
 
@@ -132,8 +124,6 @@ inline bool operator<(const AudioParameters& a, const AudioParameters& b) {
     return a.format() < b.format();
   if (a.channels() != b.channels())
     return a.channels() < b.channels();
-  if (a.input_channels() != b.input_channels())
-    return a.input_channels() < b.input_channels();
   if (a.sample_rate() != b.sample_rate())
     return a.sample_rate() < b.sample_rate();
   if (a.bits_per_sample() != b.bits_per_sample())
diff --git a/media/audio/cras/audio_manager_cras.cc b/media/audio/cras/audio_manager_cras.cc
index abc78dd..ef46176 100644
--- a/media/audio/cras/audio_manager_cras.cc
+++ b/media/audio/cras/audio_manager_cras.cc
@@ -124,12 +124,10 @@ AudioParameters AudioManagerCras::GetPreferredOutputStreamParameters(
   int sample_rate = kDefaultSampleRate;
   int buffer_size = kMinimumOutputBufferSize;
   int bits_per_sample = 16;
-  int input_channels = 0;
   if (input_params.IsValid()) {
     sample_rate = input_params.sample_rate();
     bits_per_sample = input_params.bits_per_sample();
     channel_layout = input_params.channel_layout();
-    input_channels = input_params.input_channels();
     buffer_size =
         std::min(kMaximumOutputBufferSize,
                  std::max(buffer_size, input_params.frames_per_buffer()));
@@ -140,7 +138,7 @@ AudioParameters AudioManagerCras::GetPreferredOutputStreamParameters(
     buffer_size = user_buffer_size;
 
   return AudioParameters(
-      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, input_channels,
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
       sample_rate, bits_per_sample, buffer_size, AudioParameters::NO_EFFECTS);
 }
 
diff --git a/media/audio/cras/cras_unified.cc b/media/audio/cras/cras_unified.cc
index 25af383..42d9222 100644
--- a/media/audio/cras/cras_unified.cc
+++ b/media/audio/cras/cras_unified.cc
@@ -21,30 +21,6 @@ namespace media {
 // 5) Finally Close() is called. It cleans up and notifies the audio manager,
 // which likely will destroy this object.
 //
-// For output-only streams, a unified stream is created with 0 input channels.
-//
-// Simplified data flow for unified streams:
-//
-//   +-------------+                  +------------------+
-//   | CRAS Server |                  | Chrome Client    |
-//   +------+------+    Add Stream    +---------+--------+
-//          |<----------------------------------|
-//          |                                   |
-//          |  buffer_frames captured to shm    |
-//          |---------------------------------->|
-//          |                                   |  UnifiedCallback()
-//          |                                   |  ReadWriteAudio()
-//          |                                   |
-//          |  buffer_frames written to shm     |
-//          |<----------------------------------|
-//          |                                   |
-//         ...  Repeats for each block.        ...
-//          |                                   |
-//          |                                   |
-//          |  Remove stream                    |
-//          |<----------------------------------|
-//          |                                   |
-//
 // Simplified data flow for output only streams:
 //
 //   +-------------+                  +------------------+
@@ -86,18 +62,6 @@ CrasUnifiedStream::CrasUnifiedStream(const AudioParameters& params,
   DCHECK(manager_);
   DCHECK(params_.channels()  > 0);
 
-  // Must have at least one input or output.  If there are both they must be the
-  // same.
-  int input_channels = params_.input_channels();
-
-  if (input_channels) {
-    // A unified stream for input and output.
-    DCHECK(params_.channels() == input_channels);
-    stream_direction_ = CRAS_STREAM_UNIFIED;
-    input_bus_ = AudioBus::Create(input_channels,
-                                  params_.frames_per_buffer());
-  }
-
   output_bus_ = AudioBus::Create(params);
 }
 
@@ -325,9 +289,6 @@ uint32 CrasUnifiedStream::DispatchCallback(size_t frames,
     case CRAS_STREAM_INPUT:
       NOTREACHED() << "CrasUnifiedStream doesn't support input streams.";
       return 0;
-    case CRAS_STREAM_UNIFIED:
-      return ReadWriteAudio(frames, input_samples, output_samples,
-                            input_ts, output_ts);
     default:
       break;
   }
@@ -335,37 +296,6 @@ uint32 CrasUnifiedStream::DispatchCallback(size_t frames,
   return 0;
 }
 
-// Note these are run from a real time thread, so don't waste cycles here.
-uint32 CrasUnifiedStream::ReadWriteAudio(size_t frames,
-                                         uint8* input_samples,
-                                         uint8* output_samples,
-                                         const timespec* input_ts,
-                                         const timespec* output_ts) {
-  DCHECK_EQ(frames, static_cast<size_t>(output_bus_->frames()));
-  DCHECK(source_callback_);
-
-  uint32 bytes_per_sample = bytes_per_frame_ / params_.channels();
-  input_bus_->FromInterleaved(input_samples, frames, bytes_per_sample);
-
-  // Determine latency and pass that on to the source.  We have the capture time
-  // of the first input sample and the playback time of the next audio sample
-  // passed from the audio server, add them together for total latency.
-  uint32 total_delay_bytes;
-  timespec latency_ts  = {0, 0};
-  cras_client_calc_capture_latency(input_ts, &latency_ts);
-  total_delay_bytes = GetBytesLatency(latency_ts);
-  cras_client_calc_playback_latency(output_ts, &latency_ts);
-  total_delay_bytes += GetBytesLatency(latency_ts);
-
-  int frames_filled = source_callback_->OnMoreData(
-      output_bus_.get(),
-      AudioBuffersState(0, total_delay_bytes));
-
-  output_bus_->ToInterleaved(frames_filled, bytes_per_sample, output_samples);
-
-  return frames_filled;
-}
-
 uint32 CrasUnifiedStream::WriteAudio(size_t frames,
                                      uint8* buffer,
                                      const timespec* sample_ts) {
diff --git a/media/audio/cras/cras_unified.h b/media/audio/cras/cras_unified.h
index db1d9fe..7e946e0 100644
--- a/media/audio/cras/cras_unified.h
+++ b/media/audio/cras/cras_unified.h
@@ -70,13 +70,6 @@ class MEDIA_EXPORT CrasUnifiedStream : public AudioOutputStream {
                           const timespec* input_ts,
                           const timespec* output_ts);
 
-  // Receives input samples and write output samples for a unified I/O stream.
-  uint32 ReadWriteAudio(size_t frames,
-                        uint8* input_samples,
-                        uint8* output_samples,
-                        const timespec* input_ts,
-                        const timespec* output_ts);
-
   // Writes audio for a playback stream.
   uint32 WriteAudio(size_t frames, uint8* buffer, const timespec* sample_ts);
 
@@ -108,7 +101,6 @@ class MEDIA_EXPORT CrasUnifiedStream : public AudioOutputStream {
   AudioSourceCallback* source_callback_;
 
   // Container for exchanging data with AudioSourceCallback::OnMoreData().
-  scoped_ptr<AudioBus> input_bus_;
   scoped_ptr<AudioBus> output_bus_;
 
   // Direciton of the stream.
diff --git a/media/audio/fake_audio_manager.cc b/media/audio/fake_audio_manager.cc
index e5d9bd4..a181a2f 100644
--- a/media/audio/fake_audio_manager.cc
+++ b/media/audio/fake_audio_manager.cc
@@ -58,17 +58,15 @@ AudioParameters FakeAudioManager::GetPreferredOutputStreamParameters(
   int sample_rate = kDefaultSampleRate;
   int buffer_size = kDefaultOutputBufferSize;
   int bits_per_sample = 16;
-  int input_channels = 0;
   if (input_params.IsValid()) {
     sample_rate = input_params.sample_rate();
     bits_per_sample = input_params.bits_per_sample();
     channel_layout = input_params.channel_layout();
-    input_channels = input_params.input_channels();
     buffer_size = std::min(input_params.frames_per_buffer(), buffer_size);
   }
 
   return AudioParameters(
-      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, input_channels,
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
       sample_rate, bits_per_sample, buffer_size, AudioParameters::NO_EFFECTS);
 }
 
diff --git a/media/audio/mac/audio_manager_mac.cc b/media/audio/mac/audio_manager_mac.cc
index eabce32..336a8c4 100644
--- a/media/audio/mac/audio_manager_mac.cc
+++ b/media/audio/mac/audio_manager_mac.cc
@@ -674,20 +674,9 @@ AudioParameters AudioManagerMac::GetPreferredOutputStreamParameters(
       channel_layout = CHANNEL_LAYOUT_DISCRETE;
   }
 
-  const int input_channels =
-      has_valid_input_params ? input_params.input_channels() : 0;
-  if (input_channels > 0) {
-    // TODO(xians): given the limitations of the AudioOutputStream
-    // back-ends used with synchronized I/O, we hard-code to stereo.
-    // Specifically, this is a limitation of AudioSynchronizedStream which
-    // can be removed as part of the work to consolidate these back-ends.
-    channel_layout = CHANNEL_LAYOUT_STEREO;
-  }
-
   return AudioParameters(
       AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, output_channels,
-      input_channels, hardware_sample_rate, 16, buffer_size,
-      AudioParameters::NO_EFFECTS);
+      hardware_sample_rate, 16, buffer_size, AudioParameters::NO_EFFECTS);
 }
 
 void AudioManagerMac::InitializeOnAudioThread() {
diff --git a/media/audio/openbsd/audio_manager_openbsd.cc b/media/audio/openbsd/audio_manager_openbsd.cc
index ec482d8..618fb0c 100644
--- a/media/audio/openbsd/audio_manager_openbsd.cc
+++ b/media/audio/openbsd/audio_manager_openbsd.cc
@@ -127,12 +127,10 @@ AudioParameters AudioManagerOpenBSD::GetPreferredOutputStreamParameters(
   int sample_rate = kDefaultSampleRate;
   int buffer_size = kDefaultOutputBufferSize;
   int bits_per_sample = 16;
-  int input_channels = 0;
   if (input_params.IsValid()) {
     sample_rate = input_params.sample_rate();
     bits_per_sample = input_params.bits_per_sample();
     channel_layout = input_params.channel_layout();
-    input_channels = input_params.input_channels();
     buffer_size = std::min(buffer_size, input_params.frames_per_buffer());
   }
 
@@ -141,7 +139,7 @@ AudioParameters AudioManagerOpenBSD::GetPreferredOutputStreamParameters(
     buffer_size = user_buffer_size;
 
   return AudioParameters(
-      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, input_channels,
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
       sample_rate, bits_per_sample, buffer_size, AudioParameters::NO_EFFECTS);
 }
 
diff --git a/media/audio/pulse/audio_manager_pulse.cc b/media/audio/pulse/audio_manager_pulse.cc
index 03ff17d..c752a9e 100644
--- a/media/audio/pulse/audio_manager_pulse.cc
+++ b/media/audio/pulse/audio_manager_pulse.cc
@@ -176,12 +176,10 @@ AudioParameters AudioManagerPulse::GetPreferredOutputStreamParameters(
   ChannelLayout channel_layout = CHANNEL_LAYOUT_STEREO;
   int buffer_size = kMinimumOutputBufferSize;
   int bits_per_sample = 16;
-  int input_channels = 0;
   int sample_rate = GetNativeSampleRate();
   if (input_params.IsValid()) {
     bits_per_sample = input_params.bits_per_sample();
     channel_layout = input_params.channel_layout();
-    input_channels = input_params.input_channels();
     buffer_size =
         std::min(kMaximumOutputBufferSize,
                  std::max(buffer_size, input_params.frames_per_buffer()));
@@ -192,7 +190,7 @@ AudioParameters AudioManagerPulse::GetPreferredOutputStreamParameters(
     buffer_size = user_buffer_size;
 
   return AudioParameters(
-      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, input_channels,
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
       sample_rate, bits_per_sample, buffer_size, AudioParameters::NO_EFFECTS);
 }
 
diff --git a/media/audio/win/audio_low_latency_input_win.cc b/media/audio/win/audio_low_latency_input_win.cc
index d29d1b4..29a2e7d 100644
--- a/media/audio/win/audio_low_latency_input_win.cc
+++ b/media/audio/win/audio_low_latency_input_win.cc
@@ -282,7 +282,7 @@ AudioParameters WASAPIAudioInputStream::GetInputStreamParameters(
   // Use 10ms frame size as default.
   int frames_per_buffer = sample_rate / 100;
   return AudioParameters(
-      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, 0, sample_rate,
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, sample_rate,
       16, frames_per_buffer, effects);
 }
 
diff --git a/media/audio/win/audio_low_latency_input_win_unittest.cc b/media/audio/win/audio_low_latency_input_win_unittest.cc
index eee1887..9ce7acf 100644
--- a/media/audio/win/audio_low_latency_input_win_unittest.cc
+++ b/media/audio/win/audio_low_latency_input_win_unittest.cc
@@ -208,7 +208,6 @@ class AudioInputStreamWrapper {
   AudioInputStream* CreateInputStream() {
     AudioInputStream* ais = audio_man_->MakeAudioInputStream(
         AudioParameters(format(), default_params_.channel_layout(),
-                        default_params_.input_channels(),
                         sample_rate(), bits_per_sample(), frames_per_buffer_,
                         default_params_.effects()),
         AudioManagerBase::kDefaultDeviceId);
diff --git a/media/audio/win/audio_manager_win.cc b/media/audio/win/audio_manager_win.cc
index eb05ca0..b4e5e77 100644
--- a/media/audio/win/audio_manager_win.cc
+++ b/media/audio/win/audio_manager_win.cc
@@ -305,7 +305,7 @@ AudioParameters AudioManagerWin::GetInputStreamParameters(
   if (!core_audio_supported()) {
     // Windows Wave implementation is being used.
     parameters = AudioParameters(
-        AudioParameters::AUDIO_PCM_LINEAR, CHANNEL_LAYOUT_STEREO, 0, 48000,
+        AudioParameters::AUDIO_PCM_LINEAR, CHANNEL_LAYOUT_STEREO, 48000,
         16, kFallbackBufferSize, AudioParameters::NO_EFFECTS);
   } else  {
     parameters = WASAPIAudioInputStream::GetInputStreamParameters(device_id);
@@ -314,9 +314,8 @@ AudioParameters AudioManagerWin::GetInputStreamParameters(
   int user_buffer_size = GetUserBufferSize();
   if (user_buffer_size) {
     parameters.Reset(parameters.format(), parameters.channel_layout(),
-                     parameters.channels(), parameters.input_channels(),
-                     parameters.sample_rate(), parameters.bits_per_sample(),
-                     user_buffer_size);
+                     parameters.channels(), parameters.sample_rate(),
+                     parameters.bits_per_sample(), user_buffer_size);
   }
 
   return parameters;
@@ -422,7 +421,6 @@ AudioParameters AudioManagerWin::GetPreferredOutputStreamParameters(
   int sample_rate = 48000;
   int buffer_size = kFallbackBufferSize;
   int bits_per_sample = 16;
-  int input_channels = 0;
   int effects = AudioParameters::NO_EFFECTS;
   bool use_input_params = !core_audio_supported();
   if (core_audio_supported()) {
@@ -487,7 +485,7 @@ AudioParameters AudioManagerWin::GetPreferredOutputStreamParameters(
         }
       }
     }
-    input_channels = input_params.input_channels();
+
     effects |= input_params.effects();
     if (use_input_params) {
       // If WASAPI isn't supported we'll fallback to WaveOut, which will take
@@ -507,7 +505,7 @@ AudioParameters AudioManagerWin::GetPreferredOutputStreamParameters(
     buffer_size = user_buffer_size;
 
   return AudioParameters(
-      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, input_channels,
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
       sample_rate, bits_per_sample, buffer_size, effects);
 }
 
diff --git a/media/audio/win/core_audio_util_win.cc b/media/audio/win/core_audio_util_win.cc
index 29e955ef..a01229c 100644
--- a/media/audio/win/core_audio_util_win.cc
+++ b/media/audio/win/core_audio_util_win.cc
@@ -698,8 +698,8 @@ HRESULT CoreAudioUtil::GetPreferredAudioParameters(
   if (role == eCommunications) {
     // Raise the 'DUCKING' flag for default communication devices.
     *params = AudioParameters(params->format(), params->channel_layout(),
-        params->channels(), params->input_channels(), params->sample_rate(),
-        params->bits_per_sample(), params->frames_per_buffer(),
+        params->channels(), params->sample_rate(), params->bits_per_sample(),
+        params->frames_per_buffer(),
         params->effects() | AudioParameters::DUCKING);
   }