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authorGlenn Kasten <gkasten@google.com>2015-01-26 16:35:47 -0800
committerGlenn Kasten <gkasten@google.com>2015-03-13 16:22:16 -0700
commitf59497bd3c190e087202043de5450ef06e92b27d (patch)
tree52ff88a5a4bd358358850e0ca16d0f9527fded4e /include
parent802a568f77b9a372537e4216b2a7cbb203958a87 (diff)
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Change-Id: I37d3c4ce22b74fe8581a886fe5a7f9fef8266dad
Diffstat (limited to 'include')
-rw-r--r--include/media/AudioRecord.h4
-rw-r--r--include/media/AudioSystem.h4
-rw-r--r--include/media/IAudioFlinger.h2
-rw-r--r--include/private/media/AudioTrackShared.h4
4 files changed, 8 insertions, 6 deletions
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index f70d981..59bfc4e 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -88,8 +88,8 @@ public:
* user: Pointer to context for use by the callback receiver.
* info: Pointer to optional parameter according to event type:
* - 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
- * consumed.
+ * more bytes than indicated by 'size' field and update 'size' if
+ * fewer bytes are consumed.
* - EVENT_OVERRUN: unused.
* - 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.
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 2ab3dd6..5d57c41 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -110,7 +110,7 @@ public:
// audio_stream->get_buffer_size()/audio_stream_out_frame_size()
static status_t getFrameCount(audio_io_handle_t output,
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()
static status_t getLatency(audio_io_handle_t output,
uint32_t* latency);
@@ -122,7 +122,7 @@ public:
static status_t setVoiceVolume(float volume);
// 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:
// - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
// - INVALID_OPERATION: Not supported on current hardware platform
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 31a14f0..397366b 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -94,6 +94,8 @@ public:
sp<IMemory>& buffers, // return value 0 means it follows cblk
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,
* and therefore can be cached.
*/
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 7143f1a..5644428 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -53,8 +53,8 @@ namespace android {
struct AudioTrackSharedStreaming {
// similar to NBAIO MonoPipe
// 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 mRear; // write by client
+ volatile int32_t mFront; // read by consumer (output: server, input: 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;
// server notices and discards all data between mFront and mRear
volatile uint32_t mUnderrunFrames; // server increments for each unavailable but desired frame