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Cleanup of iOS AudioDevice implementation

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TBR=tkchin
BUG=webrtc:4789
TEST=modules_unittests --gtest_filter=AudioDeviceTest* and AppRTCDemo

Review URL: https://codereview.webrtc.org/1206783002 .

Cr-Commit-Position: refs/heads/master@{#9578}
This commit is contained in:
henrika
2015-07-14 17:04:08 +02:00
parent d6f1a38165
commit ba35d05a49
18 changed files with 2790 additions and 2388 deletions
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237
webrtc/modules/audio_device/include/audio_device_defines.h
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@ -26,113 +26,164 @@ static const int kAdmMaxPlayoutBufferSizeMs = 250;
// AudioDeviceObserver
// ----------------------------------------------------------------------------
class AudioDeviceObserver
{
public:
enum ErrorCode
{
kRecordingError = 0,
kPlayoutError = 1
};
enum WarningCode
{
kRecordingWarning = 0,
kPlayoutWarning = 1
};
class AudioDeviceObserver {
public:
enum ErrorCode { kRecordingError = 0, kPlayoutError = 1 };
enum WarningCode { kRecordingWarning = 0, kPlayoutWarning = 1 };
virtual void OnErrorIsReported(const ErrorCode error) = 0;
virtual void OnWarningIsReported(const WarningCode warning) = 0;
virtual void OnErrorIsReported(const ErrorCode error) = 0;
virtual void OnWarningIsReported(const WarningCode warning) = 0;
protected:
virtual ~AudioDeviceObserver() {}
protected:
virtual ~AudioDeviceObserver() {}
};
// ----------------------------------------------------------------------------
// AudioTransport
// ----------------------------------------------------------------------------
class AudioTransport
{
public:
virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
const uint32_t nSamples,
const uint8_t nBytesPerSample,
const uint8_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
const uint32_t currentMicLevel,
const bool keyPressed,
uint32_t& newMicLevel) = 0;
class AudioTransport {
public:
virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
const uint32_t nSamples,
const uint8_t nBytesPerSample,
const uint8_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
const uint32_t currentMicLevel,
const bool keyPressed,
uint32_t& newMicLevel) = 0;
virtual int32_t NeedMorePlayData(const uint32_t nSamples,
const uint8_t nBytesPerSample,
const uint8_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
uint32_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) = 0;
virtual int32_t NeedMorePlayData(const uint32_t nSamples,
const uint8_t nBytesPerSample,
const uint8_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
uint32_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) = 0;
// Method to pass captured data directly and unmixed to network channels.
// |channel_ids| contains a list of VoE channels which are the
// sinks to the capture data. |audio_delay_milliseconds| is the sum of
// recording delay and playout delay of the hardware. |current_volume| is
// in the range of [0, 255], representing the current microphone analog
// volume. |key_pressed| is used by the typing detection.
// |need_audio_processing| specify if the data needs to be processed by APM.
// Currently WebRtc supports only one APM, and Chrome will make sure only
// one stream goes through APM. When |need_audio_processing| is false, the
// values of |audio_delay_milliseconds|, |current_volume| and |key_pressed|
// will be ignored.
// The return value is the new microphone volume, in the range of |0, 255].
// When the volume does not need to be updated, it returns 0.
// TODO(xians): Remove this interface after Chrome and Libjingle switches
// to OnData().
virtual int OnDataAvailable(const int voe_channels[],
int number_of_voe_channels,
const int16_t* audio_data,
int sample_rate,
int number_of_channels,
int number_of_frames,
int audio_delay_milliseconds,
int current_volume,
bool key_pressed,
bool need_audio_processing) { return 0; }
// Method to pass captured data directly and unmixed to network channels.
// |channel_ids| contains a list of VoE channels which are the
// sinks to the capture data. |audio_delay_milliseconds| is the sum of
// recording delay and playout delay of the hardware. |current_volume| is
// in the range of [0, 255], representing the current microphone analog
// volume. |key_pressed| is used by the typing detection.
// |need_audio_processing| specify if the data needs to be processed by APM.
// Currently WebRtc supports only one APM, and Chrome will make sure only
// one stream goes through APM. When |need_audio_processing| is false, the
// values of |audio_delay_milliseconds|, |current_volume| and |key_pressed|
// will be ignored.
// The return value is the new microphone volume, in the range of |0, 255].
// When the volume does not need to be updated, it returns 0.
// TODO(xians): Remove this interface after Chrome and Libjingle switches
// to OnData().
virtual int OnDataAvailable(const int voe_channels[],
int number_of_voe_channels,
const int16_t* audio_data,
int sample_rate,
int number_of_channels,
int number_of_frames,
int audio_delay_milliseconds,
int current_volume,
bool key_pressed,
bool need_audio_processing) {
return 0;
}
// Method to pass the captured audio data to the specific VoE channel.
// |voe_channel| is the id of the VoE channel which is the sink to the
// capture data.
// TODO(xians): Remove this interface after Libjingle switches to
// PushCaptureData().
virtual void OnData(int voe_channel, const void* audio_data,
int bits_per_sample, int sample_rate,
int number_of_channels,
int number_of_frames) {}
// Method to pass the captured audio data to the specific VoE channel.
// |voe_channel| is the id of the VoE channel which is the sink to the
// capture data.
// TODO(xians): Remove this interface after Libjingle switches to
// PushCaptureData().
virtual void OnData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
int number_of_channels,
int number_of_frames) {}
// Method to push the captured audio data to the specific VoE channel.
// The data will not undergo audio processing.
// |voe_channel| is the id of the VoE channel which is the sink to the
// capture data.
// TODO(xians): Make the interface pure virtual after Libjingle
// has its implementation.
virtual void PushCaptureData(int voe_channel, const void* audio_data,
int bits_per_sample, int sample_rate,
int number_of_channels,
int number_of_frames) {}
// Method to push the captured audio data to the specific VoE channel.
// The data will not undergo audio processing.
// |voe_channel| is the id of the VoE channel which is the sink to the
// capture data.
// TODO(xians): Make the interface pure virtual after Libjingle
// has its implementation.
virtual void PushCaptureData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
int number_of_channels,
int number_of_frames) {}
// Method to pull mixed render audio data from all active VoE channels.
// The data will not be passed as reference for audio processing internally.
// TODO(xians): Support getting the unmixed render data from specific VoE
// channel.
virtual void PullRenderData(int bits_per_sample, int sample_rate,
int number_of_channels, int number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {}
// Method to pull mixed render audio data from all active VoE channels.
// The data will not be passed as reference for audio processing internally.
// TODO(xians): Support getting the unmixed render data from specific VoE
// channel.
virtual void PullRenderData(int bits_per_sample,
int sample_rate,
int number_of_channels,
int number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {}
protected:
virtual ~AudioTransport() {}
protected:
virtual ~AudioTransport() {}
};
// Helper class for storage of fundamental audio parameters such as sample rate,
// number of channels, native buffer size etc.
// Note that one audio frame can contain more than one channel sample and each
// sample is assumed to be a 16-bit PCM sample. Hence, one audio frame in
// stereo contains 2 * (16/8) = 4 bytes of data.
class AudioParameters {
public:
// This implementation does only support 16-bit PCM samples.
enum { kBitsPerSample = 16 };
AudioParameters()
: sample_rate_(0),
channels_(0),
frames_per_buffer_(0),
frames_per_10ms_buffer_(0) {}
AudioParameters(int sample_rate, int channels, int frames_per_buffer)
: sample_rate_(sample_rate),
channels_(channels),
frames_per_buffer_(frames_per_buffer),
frames_per_10ms_buffer_(sample_rate / 100) {}
void reset(int sample_rate, int channels, int frames_per_buffer) {
sample_rate_ = sample_rate;
channels_ = channels;
frames_per_buffer_ = frames_per_buffer;
frames_per_10ms_buffer_ = (sample_rate / 100);
}
int bits_per_sample() const { return kBitsPerSample; }
int sample_rate() const { return sample_rate_; }
int channels() const { return channels_; }
int frames_per_buffer() const { return frames_per_buffer_; }
int frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; }
bool is_valid() const {
return ((sample_rate_ > 0) && (channels_ > 0) && (frames_per_buffer_ > 0));
}
int GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; }
int GetBytesPerBuffer() const {
return frames_per_buffer_ * GetBytesPerFrame();
}
int GetBytesPer10msBuffer() const {
return frames_per_10ms_buffer_ * GetBytesPerFrame();
}
float GetBufferSizeInMilliseconds() const {
if (sample_rate_ == 0)
return 0.0f;
return frames_per_buffer_ / (sample_rate_ / 1000.0f);
}
private:
int sample_rate_;
int channels_;
int frames_per_buffer_;
int frames_per_10ms_buffer_;
};
} // namespace webrtc