Chromium Code Reviews| Index: webrtc/modules/audio_device/include/audio_device_defines.h |
| diff --git a/webrtc/modules/audio_device/include/audio_device_defines.h b/webrtc/modules/audio_device/include/audio_device_defines.h |
| index 3ebbd23cc5f664d3fda73c697cbae7d53d4bc77a..b847729f05a6fa543ddaf3bd8056e9af909200d6 100644 |
| --- a/webrtc/modules/audio_device/include/audio_device_defines.h |
| +++ b/webrtc/modules/audio_device/include/audio_device_defines.h |
| @@ -49,7 +49,7 @@ class AudioTransport { |
| virtual int32_t RecordedDataIsAvailable(const void* audioSamples, |
| const size_t nSamples, |
| const size_t nBytesPerSample, |
| - const uint8_t nChannels, |
| + const size_t nChannels, |
| const uint32_t samplesPerSec, |
| const uint32_t totalDelayMS, |
| const int32_t clockDrift, |
| @@ -59,7 +59,7 @@ class AudioTransport { |
| virtual int32_t NeedMorePlayData(const size_t nSamples, |
| const size_t nBytesPerSample, |
| - const uint8_t nChannels, |
| + const size_t nChannels, |
| const uint32_t samplesPerSec, |
| void* audioSamples, |
| size_t& nSamplesOut, |
| @@ -82,10 +82,10 @@ class AudioTransport { |
| // TODO(xians): Remove this interface after Chrome and Libjingle switches |
| // to OnData(). |
| virtual int OnDataAvailable(const int voe_channels[], |
| - int number_of_voe_channels, |
| + size_t number_of_voe_channels, |
| const int16_t* audio_data, |
| int sample_rate, |
| - int number_of_channels, |
| + size_t number_of_channels, |
| size_t number_of_frames, |
| int audio_delay_milliseconds, |
| int current_volume, |
| @@ -103,7 +103,7 @@ class AudioTransport { |
| const void* audio_data, |
| int bits_per_sample, |
| int sample_rate, |
| - int number_of_channels, |
| + size_t number_of_channels, |
| size_t number_of_frames) {} |
| // Method to push the captured audio data to the specific VoE channel. |
| @@ -116,7 +116,7 @@ class AudioTransport { |
| const void* audio_data, |
| int bits_per_sample, |
| int sample_rate, |
| - int number_of_channels, |
| + size_t number_of_channels, |
| size_t number_of_frames) {} |
| // Method to pull mixed render audio data from all active VoE channels. |
| @@ -125,7 +125,7 @@ class AudioTransport { |
| // channel. |
| virtual void PullRenderData(int bits_per_sample, |
| int sample_rate, |
| - int number_of_channels, |
| + size_t number_of_channels, |
|
henrika_webrtc
2015/12/29 09:00:57
I think this change will break Chrome here:
https
|
| size_t number_of_frames, |
| void* audio_data, |
| int64_t* elapsed_time_ms, |
| @@ -149,27 +149,27 @@ class AudioParameters { |
| channels_(0), |
| frames_per_buffer_(0), |
| frames_per_10ms_buffer_(0) {} |
| - AudioParameters(int sample_rate, int channels, size_t frames_per_buffer) |
| + AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer) |
| : sample_rate_(sample_rate), |
| channels_(channels), |
| frames_per_buffer_(frames_per_buffer), |
| frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {} |
| - void reset(int sample_rate, int channels, size_t frames_per_buffer) { |
| + void reset(int sample_rate, size_t channels, size_t frames_per_buffer) { |
| sample_rate_ = sample_rate; |
| channels_ = channels; |
| frames_per_buffer_ = frames_per_buffer; |
| frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100); |
| } |
| size_t bits_per_sample() const { return kBitsPerSample; } |
| - void reset(int sample_rate, int channels, double ms_per_buffer) { |
| + void reset(int sample_rate, size_t channels, double ms_per_buffer) { |
| reset(sample_rate, channels, |
| static_cast<size_t>(sample_rate * ms_per_buffer + 0.5)); |
| } |
| - void reset(int sample_rate, int channels) { |
| + void reset(int sample_rate, size_t channels) { |
| reset(sample_rate, channels, static_cast<size_t>(0)); |
| } |
| int sample_rate() const { return sample_rate_; } |
| - int channels() const { return channels_; } |
| + size_t channels() const { return channels_; } |
| size_t frames_per_buffer() const { return frames_per_buffer_; } |
| size_t frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; } |
| size_t GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; } |
| @@ -200,7 +200,7 @@ class AudioParameters { |
| private: |
| int sample_rate_; |
| - int channels_; |
| + size_t channels_; |
| size_t frames_per_buffer_; |
| size_t frames_per_10ms_buffer_; |
| }; |