audio: different frame size calculation for input and output
Bug: 15000850.
Change-Id: I7813e99a0b7ce613cc3b7d7c95be0525cb2d6c81
diff --git a/modules/audio_remote_submix/audio_hw.cpp b/modules/audio_remote_submix/audio_hw.cpp
index 2a97347..c7e4305 100644
--- a/modules/audio_remote_submix/audio_hw.cpp
+++ b/modules/audio_remote_submix/audio_hw.cpp
@@ -416,8 +416,8 @@
device_config->buffer_size_frames = sink->maxFrames();
device_config->buffer_period_size_frames = device_config->buffer_size_frames /
buffer_period_count;
- if (in) device_config->pipe_frame_size = audio_stream_frame_size(&in->stream.common);
- if (out) device_config->pipe_frame_size = audio_stream_frame_size(&out->stream.common);
+ if (in) device_config->pipe_frame_size = audio_stream_in_frame_size(&in->stream);
+ if (out) device_config->pipe_frame_size = audio_stream_out_frame_size(&out->stream);
#if ENABLE_CHANNEL_CONVERSION
// Calculate the pipe frame size based upon the number of channels.
device_config->pipe_frame_size = (device_config->pipe_frame_size * pipe_channel_count) /
@@ -527,9 +527,9 @@
// Calculate the maximum size of the pipe buffer in frames for the specified stream.
static size_t calculate_stream_pipe_size_in_frames(const struct audio_stream *stream,
const struct submix_config *config,
- const size_t pipe_frames)
+ const size_t pipe_frames,
+ const size_t stream_frame_size)
{
- const size_t stream_frame_size = audio_stream_frame_size(stream);
const size_t pipe_frame_size = config->pipe_frame_size;
const size_t max_frame_size = max(stream_frame_size, pipe_frame_size);
return (pipe_frames * config->pipe_frame_size) / max_frame_size;
@@ -557,7 +557,7 @@
// The sample rate of the stream can't be changed once it's set since this would change the
// output buffer size and hence break playback to the shared pipe.
if (rate != out->dev->config.output_sample_rate) {
- ALOGE("out_set_sample_rate(rate=%u) resampling enabled can't change sample rate from "
+ ALOGE("out_set_sample_rate() resampling enabled can't change sample rate from "
"%u to %u", out->dev->config.output_sample_rate, rate);
return -ENOSYS;
}
@@ -576,9 +576,11 @@
const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
const_cast<struct audio_stream *>(stream));
const struct submix_config * const config = &out->dev->config;
+ const size_t stream_frame_size =
+ audio_stream_out_frame_size((const struct audio_stream_out *)stream);
const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
- stream, config, config->buffer_period_size_frames);
- const size_t buffer_size_bytes = buffer_size_frames * audio_stream_frame_size(stream);
+ stream, config, config->buffer_period_size_frames, stream_frame_size);
+ const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size;
SUBMIX_ALOGV("out_get_buffer_size() returns %zu bytes, %zu frames",
buffer_size_bytes, buffer_size_frames);
return buffer_size_bytes;
@@ -673,8 +675,10 @@
const struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(
const_cast<struct audio_stream_out *>(stream));
const struct submix_config * const config = &out->dev->config;
+ const size_t stream_frame_size =
+ audio_stream_out_frame_size(stream);
const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
- &stream->common, config, config->buffer_size_frames);
+ &stream->common, config, config->buffer_size_frames, stream_frame_size);
const uint32_t sample_rate = out_get_sample_rate(&stream->common);
const uint32_t latency_ms = (buffer_size_frames * 1000) / sample_rate;
SUBMIX_ALOGV("out_get_latency() returns %u ms, size in frames %zu, sample rate %u",
@@ -696,7 +700,7 @@
{
SUBMIX_ALOGV("out_write(bytes=%zd)", bytes);
ssize_t written_frames = 0;
- const size_t frame_size = audio_stream_frame_size(&stream->common);
+ const size_t frame_size = audio_stream_out_frame_size(stream);
struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream);
struct submix_audio_device * const rsxadev = out->dev;
const size_t frames = bytes / frame_size;
@@ -831,7 +835,7 @@
// The sample rate of the stream can't be changed once it's set since this would change the
// input buffer size and hence break recording from the shared pipe.
if (rate != in->dev->config.input_sample_rate) {
- ALOGE("in_set_sample_rate(rate=%u) resampling enabled can't change sample rate from "
+ ALOGE("in_set_sample_rate() resampling enabled can't change sample rate from "
"%u to %u", in->dev->config.input_sample_rate, rate);
return -ENOSYS;
}
@@ -850,8 +854,10 @@
const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
const_cast<struct audio_stream*>(stream));
const struct submix_config * const config = &in->dev->config;
+ const size_t stream_frame_size =
+ audio_stream_in_frame_size((const struct audio_stream_in *)stream);
size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
- stream, config, config->buffer_period_size_frames);
+ stream, config, config->buffer_period_size_frames, stream_frame_size);
#if ENABLE_RESAMPLING
// Scale the size of the buffer based upon the maximum number of frames that could be returned
// given the ratio of output to input sample rate.
@@ -859,7 +865,7 @@
(float)config->input_sample_rate) /
(float)config->output_sample_rate);
#endif // ENABLE_RESAMPLING
- const size_t buffer_size_bytes = buffer_size_frames * audio_stream_frame_size(stream);
+ const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size;
SUBMIX_ALOGV("in_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes,
buffer_size_frames);
return buffer_size_bytes;
@@ -943,7 +949,7 @@
struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream);
struct submix_audio_device * const rsxadev = in->dev;
struct audio_config *format;
- const size_t frame_size = audio_stream_frame_size(&stream->common);
+ const size_t frame_size = audio_stream_in_frame_size(stream);
const size_t frames_to_read = bytes / frame_size;
SUBMIX_ALOGV("in_read bytes=%zu", bytes);