Merge "r_submix HAL: remove legacy code for in-pipe conversions"
diff --git a/modules/audio_remote_submix/audio_hw.cpp b/modules/audio_remote_submix/audio_hw.cpp
index b43a44d..a944caa 100644
--- a/modules/audio_remote_submix/audio_hw.cpp
+++ b/modules/audio_remote_submix/audio_hw.cpp
@@ -83,10 +83,7 @@
// multiple input streams from this device. If this option is enabled, each input stream returned
// is *the same stream* which means that readers will race to read data from these streams.
#define ENABLE_LEGACY_INPUT_OPEN 1
-// Whether channel conversion (16-bit signed PCM mono->stereo, stereo->mono) is enabled.
-#define ENABLE_CHANNEL_CONVERSION 1
-// Whether resampling is enabled.
-#define ENABLE_RESAMPLING 1
+
#if LOG_STREAMS_TO_FILES
// Folder to save stream log files to.
#define LOG_STREAM_FOLDER "/data/misc/audioserver"
@@ -130,11 +127,6 @@
// channel bitfields are not equivalent.
audio_channel_mask_t input_channel_mask;
audio_channel_mask_t output_channel_mask;
-#if ENABLE_RESAMPLING
- // Input stream and output stream sample rates.
- uint32_t input_sample_rate;
- uint32_t output_sample_rate;
-#endif // ENABLE_RESAMPLING
size_t pipe_frame_size; // Number of bytes in each audio frame in the pipe.
size_t buffer_size_frames; // Size of the audio pipe in frames.
// Maximum number of frames buffered by the input and output streams.
@@ -159,11 +151,6 @@
// destroyed if both and input and output streams are destroyed.
struct submix_stream_out *output;
struct submix_stream_in *input;
-#if ENABLE_RESAMPLING
- // Buffer used as temporary storage for resampled data prior to returning data to the output
- // stream.
- int16_t resampler_buffer[DEFAULT_PIPE_SIZE_IN_FRAMES];
-#endif // ENABLE_RESAMPLING
} route_config_t;
struct submix_audio_device {
@@ -325,7 +312,6 @@
static bool audio_config_compare(const audio_config * const input_config,
const audio_config * const output_config)
{
-#if !ENABLE_CHANNEL_CONVERSION
const uint32_t input_channels = audio_channel_count_from_in_mask(input_config->channel_mask);
const uint32_t output_channels = audio_channel_count_from_out_mask(output_config->channel_mask);
if (input_channels != output_channels) {
@@ -333,13 +319,8 @@
input_channels, output_channels);
return false;
}
-#endif // !ENABLE_CHANNEL_CONVERSION
-#if ENABLE_RESAMPLING
- if (input_config->sample_rate != output_config->sample_rate &&
- audio_channel_count_from_in_mask(input_config->channel_mask) != 1) {
-#else
+
if (input_config->sample_rate != output_config->sample_rate) {
-#endif // ENABLE_RESAMPLING
ALOGE("audio_config_compare() sample rate mismatch %ul vs. %ul",
input_config->sample_rate, output_config->sample_rate);
return false;
@@ -376,24 +357,11 @@
in->route_handle = route_idx;
rsxadev->routes[route_idx].input = in;
rsxadev->routes[route_idx].config.input_channel_mask = config->channel_mask;
-#if ENABLE_RESAMPLING
- rsxadev->routes[route_idx].config.input_sample_rate = config->sample_rate;
- // If the output isn't configured yet, set the output sample rate to the maximum supported
- // sample rate such that the smallest possible input buffer is created, and put a default
- // value for channel count
- if (!rsxadev->routes[route_idx].output) {
- rsxadev->routes[route_idx].config.output_sample_rate = 48000;
- rsxadev->routes[route_idx].config.output_channel_mask = AUDIO_CHANNEL_OUT_STEREO;
- }
-#endif // ENABLE_RESAMPLING
}
if (out) {
out->route_handle = route_idx;
rsxadev->routes[route_idx].output = out;
rsxadev->routes[route_idx].config.output_channel_mask = config->channel_mask;
-#if ENABLE_RESAMPLING
- rsxadev->routes[route_idx].config.output_sample_rate = config->sample_rate;
-#endif // ENABLE_RESAMPLING
}
// Save the address
strncpy(rsxadev->routes[route_idx].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN);
@@ -403,18 +371,14 @@
{
struct submix_config * const device_config = &rsxadev->routes[route_idx].config;
uint32_t channel_count;
- if (out)
+ if (out) {
channel_count = audio_channel_count_from_out_mask(config->channel_mask);
- else
+ } else {
channel_count = audio_channel_count_from_in_mask(config->channel_mask);
-#if ENABLE_CHANNEL_CONVERSION
- // If channel conversion is enabled, allocate enough space for the maximum number of
- // possible channels stored in the pipe for the situation when the number of channels in
- // the output stream don't match the number in the input stream.
- const uint32_t pipe_channel_count = max(channel_count, 2);
-#else
+ }
+
const uint32_t pipe_channel_count = channel_count;
-#endif // ENABLE_CHANNEL_CONVERSION
+
const NBAIO_Format format = Format_from_SR_C(config->sample_rate, pipe_channel_count,
config->format);
const NBAIO_Format offers[1] = {format};
@@ -444,11 +408,7 @@
buffer_period_count;
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) /
- channel_count;
-#endif // ENABLE_CHANNEL_CONVERSION
+
SUBMIX_ALOGV("submix_audio_device_create_pipe_l(): pipe frame size %zd, pipe size %zd, "
"period size %zd", device_config->pipe_frame_size,
device_config->buffer_size_frames, device_config->buffer_period_size_frames);
@@ -473,10 +433,6 @@
rsxadev->routes[route_idx].rsxSource.clear();
}
memset(rsxadev->routes[route_idx].address, 0, AUDIO_DEVICE_MAX_ADDRESS_LEN);
-#if ENABLE_RESAMPLING
- memset(rsxadev->routes[route_idx].resampler_buffer, 0,
- sizeof(int16_t) * DEFAULT_PIPE_SIZE_IN_FRAMES);
-#endif
}
// Remove references to the specified input and output streams. When the device no longer
@@ -624,11 +580,7 @@
{
const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
const_cast<struct audio_stream *>(stream));
-#if ENABLE_RESAMPLING
- const uint32_t out_rate = out->dev->routes[out->route_handle].config.output_sample_rate;
-#else
const uint32_t out_rate = out->dev->routes[out->route_handle].config.common.sample_rate;
-#endif // ENABLE_RESAMPLING
SUBMIX_ALOGV("out_get_sample_rate() returns %u for addr %s",
out_rate, out->dev->routes[out->route_handle].address);
return out_rate;
@@ -637,17 +589,6 @@
static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream);
-#if ENABLE_RESAMPLING
- // 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->routes[out->route_handle].config.output_sample_rate) {
- ALOGE("out_set_sample_rate() resampling enabled can't change sample rate from "
- "%u to %u for addr %s",
- out->dev->routes[out->route_handle].config.output_sample_rate, rate,
- out->dev->routes[out->route_handle].address);
- return -ENOSYS;
- }
-#endif // ENABLE_RESAMPLING
if (!sample_rate_supported(rate)) {
ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate);
return -ENOSYS;
@@ -994,11 +935,7 @@
{
const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
const_cast<struct audio_stream*>(stream));
-#if ENABLE_RESAMPLING
- const uint32_t rate = in->dev->routes[in->route_handle].config.input_sample_rate;
-#else
const uint32_t rate = in->dev->routes[in->route_handle].config.common.sample_rate;
-#endif // ENABLE_RESAMPLING
SUBMIX_ALOGV("in_get_sample_rate() returns %u", rate);
return rate;
}
@@ -1006,15 +943,6 @@
static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream);
-#if ENABLE_RESAMPLING
- // 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->routes[in->route_handle].config.input_sample_rate) {
- ALOGE("in_set_sample_rate() resampling enabled can't change sample rate from "
- "%u to %u", in->dev->routes[in->route_handle].config.input_sample_rate, rate);
- return -ENOSYS;
- }
-#endif // ENABLE_RESAMPLING
if (!sample_rate_supported(rate)) {
ALOGE("in_set_sample_rate(rate=%u) rate unsupported", rate);
return -ENOSYS;
@@ -1033,13 +961,6 @@
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_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.
- buffer_size_frames = (size_t)(((float)buffer_size_frames *
- (float)config->input_sample_rate) /
- (float)config->output_sample_rate);
-#endif // ENABLE_RESAMPLING
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);
@@ -1168,65 +1089,10 @@
// read the data from the pipe (it's non blocking)
int attempts = 0;
char* buff = (char*)buffer;
-#if ENABLE_CHANNEL_CONVERSION
- // Determine whether channel conversion is required.
- const uint32_t input_channels = audio_channel_count_from_in_mask(
- rsxadev->routes[in->route_handle].config.input_channel_mask);
- const uint32_t output_channels = audio_channel_count_from_out_mask(
- rsxadev->routes[in->route_handle].config.output_channel_mask);
- if (input_channels != output_channels) {
- SUBMIX_ALOGV("in_read(): %d output channels will be converted to %d "
- "input channels", output_channels, input_channels);
- // Only support 16-bit PCM channel conversion from mono to stereo or stereo to mono.
- ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format ==
- AUDIO_FORMAT_PCM_16_BIT);
- ALOG_ASSERT((input_channels == 1 && output_channels == 2) ||
- (input_channels == 2 && output_channels == 1));
- }
-#endif // ENABLE_CHANNEL_CONVERSION
-
-#if ENABLE_RESAMPLING
- const uint32_t input_sample_rate = in_get_sample_rate(&stream->common);
- const uint32_t output_sample_rate =
- rsxadev->routes[in->route_handle].config.output_sample_rate;
- const size_t resampler_buffer_size_frames =
- sizeof(rsxadev->routes[in->route_handle].resampler_buffer) /
- sizeof(rsxadev->routes[in->route_handle].resampler_buffer[0]);
- float resampler_ratio = 1.0f;
- // Determine whether resampling is required.
- if (input_sample_rate != output_sample_rate) {
- resampler_ratio = (float)output_sample_rate / (float)input_sample_rate;
- // Only support 16-bit PCM mono resampling.
- // NOTE: Resampling is performed after the channel conversion step.
- ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format ==
- AUDIO_FORMAT_PCM_16_BIT);
- ALOG_ASSERT(audio_channel_count_from_in_mask(
- rsxadev->routes[in->route_handle].config.input_channel_mask) == 1);
- }
-#endif // ENABLE_RESAMPLING
while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) {
ssize_t frames_read = -1977;
size_t read_frames = remaining_frames;
-#if ENABLE_RESAMPLING
- char* const saved_buff = buff;
- if (resampler_ratio != 1.0f) {
- // Calculate the number of frames from the pipe that need to be read to generate
- // the data for the input stream read.
- const size_t frames_required_for_resampler = (size_t)(
- (float)read_frames * (float)resampler_ratio);
- read_frames = min(frames_required_for_resampler, resampler_buffer_size_frames);
- // Read into the resampler buffer.
- buff = (char*)rsxadev->routes[in->route_handle].resampler_buffer;
- }
-#endif // ENABLE_RESAMPLING
-#if ENABLE_CHANNEL_CONVERSION
- if (output_channels == 1 && input_channels == 2) {
- // Need to read half the requested frames since the converted output
- // data will take twice the space (mono->stereo).
- read_frames /= 2;
- }
-#endif // ENABLE_CHANNEL_CONVERSION
SUBMIX_ALOGV("in_read(): frames available to read %zd", source->availableToRead());
@@ -1234,56 +1100,6 @@
SUBMIX_ALOGV("in_read(): frames read %zd", frames_read);
-#if ENABLE_CHANNEL_CONVERSION
- // Perform in-place channel conversion.
- // NOTE: In the following "input stream" refers to the data returned by this function
- // and "output stream" refers to the data read from the pipe.
- if (input_channels != output_channels && frames_read > 0) {
- int16_t *data = (int16_t*)buff;
- if (output_channels == 2 && input_channels == 1) {
- // Offset into the output stream data in samples.
- ssize_t output_stream_offset = 0;
- for (ssize_t input_stream_frame = 0; input_stream_frame < frames_read;
- input_stream_frame++, output_stream_offset += 2) {
- // Average the content from both channels.
- data[input_stream_frame] = ((int32_t)data[output_stream_offset] +
- (int32_t)data[output_stream_offset + 1]) / 2;
- }
- } else if (output_channels == 1 && input_channels == 2) {
- // Offset into the input stream data in samples.
- ssize_t input_stream_offset = (frames_read - 1) * 2;
- for (ssize_t output_stream_frame = frames_read - 1; output_stream_frame >= 0;
- output_stream_frame--, input_stream_offset -= 2) {
- const short sample = data[output_stream_frame];
- data[input_stream_offset] = sample;
- data[input_stream_offset + 1] = sample;
- }
- }
- }
-#endif // ENABLE_CHANNEL_CONVERSION
-
-#if ENABLE_RESAMPLING
- if (resampler_ratio != 1.0f) {
- SUBMIX_ALOGV("in_read(): resampling %zd frames", frames_read);
- const int16_t * const data = (int16_t*)buff;
- int16_t * const resampled_buffer = (int16_t*)saved_buff;
- // Resample with *no* filtering - if the data from the ouptut stream was really
- // sampled at a different rate this will result in very nasty aliasing.
- const float output_stream_frames = (float)frames_read;
- size_t input_stream_frame = 0;
- for (float output_stream_frame = 0.0f;
- output_stream_frame < output_stream_frames &&
- input_stream_frame < remaining_frames;
- output_stream_frame += resampler_ratio, input_stream_frame++) {
- resampled_buffer[input_stream_frame] = data[(size_t)output_stream_frame];
- }
- ALOG_ASSERT(input_stream_frame <= (ssize_t)resampler_buffer_size_frames);
- SUBMIX_ALOGV("in_read(): resampler produced %zd frames", input_stream_frame);
- frames_read = input_stream_frame;
- buff = saved_buff;
- }
-#endif // ENABLE_RESAMPLING
-
if (frames_read > 0) {
#if LOG_STREAMS_TO_FILES
if (in->log_fd >= 0) write(in->log_fd, buff, frames_read * frame_size);
@@ -1464,13 +1280,6 @@
out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
out->stream.get_presentation_position = out_get_presentation_position;
-#if ENABLE_RESAMPLING
- // Recreate the pipe with the correct sample rate so that MonoPipe.write() rate limits
- // writes correctly.
- force_pipe_creation = rsxadev->routes[route_idx].config.common.sample_rate
- != config->sample_rate;
-#endif // ENABLE_RESAMPLING
-
// If the sink has been shutdown or pipe recreation is forced (see above), delete the pipe so
// that it's recreated.
if ((rsxadev->routes[route_idx].rsxSink != NULL
@@ -1779,16 +1588,9 @@
int n = snprintf(msg, sizeof(msg), "\nReroute submix audio module:\n");
write(fd, &msg, n);
for (int i=0 ; i < MAX_ROUTES ; i++) {
-#if ENABLE_RESAMPLING
- n = snprintf(msg, sizeof(msg), " route[%d] rate in=%d out=%d, addr=[%s]\n", i,
- rsxadev->routes[i].config.input_sample_rate,
- rsxadev->routes[i].config.output_sample_rate,
- rsxadev->routes[i].address);
-#else
n = snprintf(msg, sizeof(msg), " route[%d], rate=%d addr=[%s]\n", i,
rsxadev->routes[i].config.common.sample_rate,
rsxadev->routes[i].address);
-#endif
write(fd, &msg, n);
}
return 0;