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Diffstat (limited to 'webrtc/modules/audio_device/include/audio_device_defines.h')
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1 files changed, 210 insertions, 0 deletions
diff --git a/webrtc/modules/audio_device/include/audio_device_defines.h b/webrtc/modules/audio_device/include/audio_device_defines.h new file mode 100644 index 0000000000..3ebbd23cc5 --- /dev/null +++ b/webrtc/modules/audio_device/include/audio_device_defines.h @@ -0,0 +1,210 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_ +#define WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_ + +#include <stddef.h> + +#include "webrtc/typedefs.h" + +namespace webrtc { + +static const int kAdmMaxDeviceNameSize = 128; +static const int kAdmMaxFileNameSize = 512; +static const int kAdmMaxGuidSize = 128; + +static const int kAdmMinPlayoutBufferSizeMs = 10; +static const int kAdmMaxPlayoutBufferSizeMs = 250; + +// ---------------------------------------------------------------------------- +// AudioDeviceObserver +// ---------------------------------------------------------------------------- + +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; + + protected: + virtual ~AudioDeviceObserver() {} +}; + +// ---------------------------------------------------------------------------- +// AudioTransport +// ---------------------------------------------------------------------------- + +class AudioTransport { + public: + virtual int32_t RecordedDataIsAvailable(const void* audioSamples, + const size_t nSamples, + const size_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 size_t nSamples, + const size_t nBytesPerSample, + const uint8_t nChannels, + const uint32_t samplesPerSec, + void* audioSamples, + size_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, + size_t 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, + size_t 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, + size_t 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, + size_t number_of_frames, + void* audio_data, + int64_t* elapsed_time_ms, + int64_t* ntp_time_ms) {} + + 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. + static const size_t kBitsPerSample = 16; + AudioParameters() + : sample_rate_(0), + channels_(0), + frames_per_buffer_(0), + frames_per_10ms_buffer_(0) {} + AudioParameters(int sample_rate, int 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) { + 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) { + reset(sample_rate, channels, + static_cast<size_t>(sample_rate * ms_per_buffer + 0.5)); + } + void reset(int sample_rate, int channels) { + reset(sample_rate, channels, static_cast<size_t>(0)); + } + int sample_rate() const { return sample_rate_; } + int 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; } + size_t GetBytesPerBuffer() const { + return frames_per_buffer_ * GetBytesPerFrame(); + } + // The WebRTC audio device buffer (ADB) only requires that the sample rate + // and number of channels are configured. Hence, to be "valid", only these + // two attributes must be set. + bool is_valid() const { return ((sample_rate_ > 0) && (channels_ > 0)); } + // Most platforms also require that a native buffer size is defined. + // An audio parameter instance is considered to be "complete" if it is both + // "valid" (can be used by the ADB) and also has a native frame size. + bool is_complete() const { return (is_valid() && (frames_per_buffer_ > 0)); } + size_t GetBytesPer10msBuffer() const { + return frames_per_10ms_buffer_ * GetBytesPerFrame(); + } + double GetBufferSizeInMilliseconds() const { + if (sample_rate_ == 0) + return 0.0; + return frames_per_buffer_ / (sample_rate_ / 1000.0); + } + double GetBufferSizeInSeconds() const { + if (sample_rate_ == 0) + return 0.0; + return static_cast<double>(frames_per_buffer_) / (sample_rate_); + } + + private: + int sample_rate_; + int channels_; + size_t frames_per_buffer_; + size_t frames_per_10ms_buffer_; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_ |