| // 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. |
| |
| #ifndef MEDIA_AUDIO_AUDIO_PARAMETERS_H_ |
| #define MEDIA_AUDIO_AUDIO_PARAMETERS_H_ |
| |
| #include "base/basictypes.h" |
| #include "media/base/channel_layout.h" |
| #include "media/base/media_export.h" |
| |
| namespace media { |
| |
| struct MEDIA_EXPORT AudioInputBufferParameters { |
| double volume; |
| uint32 size; |
| }; |
| |
| // Use a struct-in-struct approach to ensure that we can calculate the required |
| // size as sizeof(AudioInputBufferParameters) + #(bytes in audio buffer) without |
| // using packing. |
| struct MEDIA_EXPORT AudioInputBuffer { |
| AudioInputBufferParameters params; |
| int8 audio[1]; |
| }; |
| |
| class MEDIA_EXPORT AudioParameters { |
| public: |
| enum Format { |
| AUDIO_PCM_LINEAR = 0, // PCM is 'raw' amplitude samples. |
| AUDIO_PCM_LOW_LATENCY, // Linear PCM, low latency requested. |
| AUDIO_FAKE, // Creates a fake AudioOutputStream object. |
| AUDIO_VIRTUAL, // Creates a VirtualAudioInputStream object. |
| // Applies to input streams only. |
| AUDIO_LAST_FORMAT // Only used for validation of format. |
| }; |
| |
| enum { |
| // Telephone quality sample rate, mostly for speech-only audio. |
| kTelephoneSampleRate = 8000, |
| // CD sampling rate is 44.1 KHz or conveniently 2x2x3x3x5x5x7x7. |
| kAudioCDSampleRate = 44100, |
| }; |
| |
| AudioParameters(); |
| AudioParameters(Format format, ChannelLayout channel_layout, |
| int sample_rate, int bits_per_sample, |
| int frames_per_buffer); |
| void Reset(Format format, ChannelLayout channel_layout, |
| int sample_rate, int bits_per_sample, |
| int frames_per_buffer); |
| |
| // Checks that all values are in the expected range. All limits are specified |
| // in media::Limits. |
| bool IsValid() const; |
| |
| // Returns size of audio buffer in bytes. |
| int GetBytesPerBuffer() const; |
| |
| // Returns the number of bytes representing one second of audio. |
| int GetBytesPerSecond() const; |
| |
| // Returns the number of bytes representing a frame of audio. |
| int GetBytesPerFrame() const; |
| |
| Format format() const { return format_; } |
| ChannelLayout channel_layout() const { return channel_layout_; } |
| int sample_rate() const { return sample_rate_; } |
| int bits_per_sample() const { return bits_per_sample_; } |
| int frames_per_buffer() const { return frames_per_buffer_; } |
| int channels() const { return channels_; } |
| |
| private: |
| Format format_; // Format of the stream. |
| ChannelLayout channel_layout_; // Order of surround sound channels. |
| int sample_rate_; // Sampling frequency/rate. |
| int bits_per_sample_; // Number of bits per sample. |
| int frames_per_buffer_; // Number of frames in a buffer. |
| |
| int channels_; // Number of channels. Value set based on |
| // |channel_layout|. |
| }; |
| |
| // Comparison is useful when AudioParameters is used with std structures. |
| inline bool operator<(const AudioParameters& a, const AudioParameters& b) { |
| if (a.format() != b.format()) |
| return a.format() < b.format(); |
| if (a.channels() != b.channels()) |
| return a.channels() < b.channels(); |
| if (a.sample_rate() != b.sample_rate()) |
| return a.sample_rate() < b.sample_rate(); |
| if (a.bits_per_sample() != b.bits_per_sample()) |
| return a.bits_per_sample() < b.bits_per_sample(); |
| return a.frames_per_buffer() < b.frames_per_buffer(); |
| } |
| |
| } // namespace media |
| |
| #endif // MEDIA_AUDIO_AUDIO_PARAMETERS_H_ |