diff options
Diffstat (limited to 'include')
| -rw-r--r-- | include/media/AudioResamplerPublic.h | 19 | ||||
| -rw-r--r-- | include/media/AudioSystem.h | 17 | ||||
| -rw-r--r-- | include/media/AudioTrack.h | 24 | ||||
| -rw-r--r-- | include/media/IAudioPolicyService.h | 1 |
4 files changed, 53 insertions, 8 deletions
diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h index b705efa..0634741 100644 --- a/include/media/AudioResamplerPublic.h +++ b/include/media/AudioResamplerPublic.h @@ -17,6 +17,8 @@ #ifndef ANDROID_AUDIO_RESAMPLER_PUBLIC_H #define ANDROID_AUDIO_RESAMPLER_PUBLIC_H +#include <stdint.h> + // AUDIO_RESAMPLER_DOWN_RATIO_MAX is the maximum ratio between the original // audio sample rate and the target rate when downsampling, // as permitted in the audio framework, e.g. AudioTrack and AudioFlinger. @@ -26,6 +28,12 @@ // TODO: replace with an API #define AUDIO_RESAMPLER_DOWN_RATIO_MAX 256 +// AUDIO_RESAMPLER_UP_RATIO_MAX is the maximum suggested ratio between the original +// audio sample rate and the target rate when upsampling. It is loosely enforced by +// the system. One issue with large upsampling ratios is the approximation by +// an int32_t of the phase increments, making the resulting sample rate inexact. +#define AUDIO_RESAMPLER_UP_RATIO_MAX 65536 + // Returns the source frames needed to resample to destination frames. This is not a precise // value and depends on the resampler (and possibly how it handles rounding internally). // Nevertheless, this should be an upper bound on the requirements of the resampler. @@ -39,4 +47,15 @@ static inline size_t sourceFramesNeeded( size_t((uint64_t)dstFramesRequired * srcSampleRate / dstSampleRate + 1 + 1); } +// An upper bound for the number of destination frames possible from srcFrames +// after sample rate conversion. This may be used for buffer sizing. +static inline size_t destinationFramesPossible(size_t srcFrames, uint32_t srcSampleRate, + uint32_t dstSampleRate) { + if (srcSampleRate == dstSampleRate) { + return srcFrames; + } + uint64_t dstFrames = (uint64_t)srcFrames * dstSampleRate / srcSampleRate; + return dstFrames > 2 ? dstFrames - 2 : 0; +} + #endif // ANDROID_AUDIO_RESAMPLER_PUBLIC_H diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h index f5db1bb..3b6db8c 100644 --- a/include/media/AudioSystem.h +++ b/include/media/AudioSystem.h @@ -221,14 +221,15 @@ public: audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, const audio_offload_info_t *offloadInfo = NULL); static status_t getOutputForAttr(const audio_attributes_t *attr, - audio_io_handle_t *output, - audio_session_t session, - audio_stream_type_t *stream, - uint32_t samplingRate = 0, - audio_format_t format = AUDIO_FORMAT_DEFAULT, - audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, - audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, - const audio_offload_info_t *offloadInfo = NULL); + audio_io_handle_t *output, + audio_session_t session, + audio_stream_type_t *stream, + uint32_t samplingRate = 0, + audio_format_t format = AUDIO_FORMAT_DEFAULT, + audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, + audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, + audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE, + const audio_offload_info_t *offloadInfo = NULL); static status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session); diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index d9b7057..e7e0703 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -477,6 +477,26 @@ private: audio_io_handle_t getOutput() const; public: + /* Selects the audio device to use for output of this AudioTrack. A value of + * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. + * + * Parameters: + * The device ID of the selected device (as returned by the AudioDevicesManager API). + * + * Returned value: + * - NO_ERROR: successful operation + * TODO: what else can happen here? + */ + status_t setOutputDevice(audio_port_handle_t deviceId); + + /* Returns the ID of the audio device used for output of this AudioTrack. + * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. + * + * Parameters: + * none. + */ + audio_port_handle_t getOutputDevice(); + /* Returns the unique session ID associated with this track. * * Parameters: @@ -817,6 +837,10 @@ protected: bool mInUnderrun; // whether track is currently in underrun state uint32_t mPausedPosition; + // For Device Selection API + // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. + int mSelectedDeviceId; + private: class DeathNotifier : public IBinder::DeathRecipient { public: diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h index fecc6f1..7506153 100644 --- a/include/media/IAudioPolicyService.h +++ b/include/media/IAudioPolicyService.h @@ -66,6 +66,7 @@ public: audio_format_t format = AUDIO_FORMAT_DEFAULT, audio_channel_mask_t channelMask = 0, audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, + audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE, const audio_offload_info_t *offloadInfo = NULL) = 0; virtual status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, |