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Commit f59497bd authored by Glenn Kasten's avatar Glenn Kasten
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Change-Id: I37d3c4ce22b74fe8581a886fe5a7f9fef8266dad
parent 802a568f
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+2 −2
Original line number Original line Diff line number Diff line
@@ -88,8 +88,8 @@ public:
     * user:    Pointer to context for use by the callback receiver.
     * user:    Pointer to context for use by the callback receiver.
     * info:    Pointer to optional parameter according to event type:
     * info:    Pointer to optional parameter according to event type:
     *          - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read
     *          - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read
     *            more bytes than indicated by 'size' field and update 'size' if fewer bytes are
     *                             more bytes than indicated by 'size' field and update 'size' if
     *            consumed.
     *                             fewer bytes are consumed.
     *          - EVENT_OVERRUN: unused.
     *          - EVENT_OVERRUN: unused.
     *          - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames.
     *          - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames.
     *          - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames.
     *          - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames.
+2 −2
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@@ -110,7 +110,7 @@ public:
    // audio_stream->get_buffer_size()/audio_stream_out_frame_size()
    // audio_stream->get_buffer_size()/audio_stream_out_frame_size()
    static status_t getFrameCount(audio_io_handle_t output,
    static status_t getFrameCount(audio_io_handle_t output,
                                  size_t* frameCount);
                                  size_t* frameCount);
    // returns the audio output stream latency in ms. Corresponds to
    // returns the audio output latency in ms. Corresponds to
    // audio_stream_out->get_latency()
    // audio_stream_out->get_latency()
    static status_t getLatency(audio_io_handle_t output,
    static status_t getLatency(audio_io_handle_t output,
                               uint32_t* latency);
                               uint32_t* latency);
@@ -122,7 +122,7 @@ public:
    static status_t setVoiceVolume(float volume);
    static status_t setVoiceVolume(float volume);


    // return the number of audio frames written by AudioFlinger to audio HAL and
    // return the number of audio frames written by AudioFlinger to audio HAL and
    // audio dsp to DAC since the specified output I/O handle has exited standby.
    // audio dsp to DAC since the specified output has exited standby.
    // returned status (from utils/Errors.h) can be:
    // returned status (from utils/Errors.h) can be:
    // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
    // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
    // - INVALID_OPERATION: Not supported on current hardware platform
    // - INVALID_OPERATION: Not supported on current hardware platform
+2 −0
Original line number Original line Diff line number Diff line
@@ -94,6 +94,8 @@ public:
                                sp<IMemory>& buffers,   // return value 0 means it follows cblk
                                sp<IMemory>& buffers,   // return value 0 means it follows cblk
                                status_t *status) = 0;
                                status_t *status) = 0;


    // FIXME Surprisingly, sampleRate/format/frameCount/latency don't work for input handles

    /* query the audio hardware state. This state never changes,
    /* query the audio hardware state. This state never changes,
     * and therefore can be cached.
     * and therefore can be cached.
     */
     */
+2 −2
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@@ -53,8 +53,8 @@ namespace android {
struct AudioTrackSharedStreaming {
struct AudioTrackSharedStreaming {
    // similar to NBAIO MonoPipe
    // similar to NBAIO MonoPipe
    // in continuously incrementing frame units, take modulo buffer size, which must be a power of 2
    // in continuously incrementing frame units, take modulo buffer size, which must be a power of 2
    volatile int32_t mFront;    // read by server
    volatile int32_t mFront;    // read by consumer (output: server, input: client)
    volatile int32_t mRear;     // write by client
    volatile int32_t mRear;     // written by producer (output: client, input: server)
    volatile int32_t mFlush;    // incremented by client to indicate a request to flush;
    volatile int32_t mFlush;    // incremented by client to indicate a request to flush;
                                // server notices and discards all data between mFront and mRear
                                // server notices and discards all data between mFront and mRear
    volatile uint32_t mUnderrunFrames;  // server increments for each unavailable but desired frame
    volatile uint32_t mUnderrunFrames;  // server increments for each unavailable but desired frame
+0 −1
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@@ -423,7 +423,6 @@ status_t AudioTrackClientProxy::waitStreamEndDone(const struct timespec *request
            goto end;
            goto end;
        }
        }
        // check for obtainBuffer interrupted by client
        // check for obtainBuffer interrupted by client
        // check for obtainBuffer interrupted by client
        if (flags & CBLK_INTERRUPT) {
        if (flags & CBLK_INTERRUPT) {
            ALOGV("waitStreamEndDone() interrupted by client");
            ALOGV("waitStreamEndDone() interrupted by client");
            status = -EINTR;
            status = -EINTR;
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