r_submix HAL: remove legacy code for in-pipe conversions
Old hacky uses of this HAL required channel and sample
rate conversion, which were never expected when this HAL
is used for its real application as the backend for
dynamic audio policies. In that context conversions are
performed in audio_flinger in the mixer.
Bug: 141604269
Test: atest AudioHostTest#testTwoChannelCapturing
Change-Id: I333e981fb3db403ab2da6f77debe49bc7d132ebc
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;