Merge "Added basic resampling of submix data on read from the input stream."
diff --git a/modules/audio_remote_submix/audio_hw.cpp b/modules/audio_remote_submix/audio_hw.cpp
index 60a0649..f11b207 100644
--- a/modules/audio_remote_submix/audio_hw.cpp
+++ b/modules/audio_remote_submix/audio_hw.cpp
@@ -84,6 +84,8 @@
#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/media"
@@ -125,6 +127,11 @@
// 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.
@@ -146,6 +153,11 @@
// TV with Wifi Display capabilities), or to a wireless audio player.
sp<MonoPipe> rsxSink;
sp<MonoPipeReader> rsxSource;
+#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
// Pointers to the current input and output stream instances. rsxSink and rsxSource are
// destroyed if both and input and output streams are destroyed.
@@ -321,7 +333,12 @@
return false;
}
#endif // !ENABLE_CHANNEL_CONVERSION
+#if ENABLE_RESAMPLING
+ if (input_config->sample_rate != output_config->sample_rate &&
+ get_channel_count_from_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;
@@ -352,10 +369,21 @@
if (in) {
rsxadev->input = in;
rsxadev->config.input_channel_mask = config->channel_mask;
+#if ENABLE_RESAMPLING
+ rsxadev->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.
+ if (!rsxadev->output) {
+ rsxadev->config.output_sample_rate = 48000;
+ }
+#endif // ENABLE_RESAMPLING
}
if (out) {
rsxadev->output = out;
rsxadev->config.output_channel_mask = config->channel_mask;
+#if ENABLE_RESAMPLING
+ rsxadev->config.output_sample_rate = config->sample_rate;
+#endif // ENABLE_RESAMPLING
}
// If a pipe isn't associated with the device, create one.
if (rsxadev->rsxSink == NULL || rsxadev->rsxSource == NULL) {
@@ -507,18 +535,31 @@
{
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->config.output_sample_rate;
+#else
const uint32_t out_rate = out->dev->config.common.sample_rate;
+#endif // ENABLE_RESAMPLING
SUBMIX_ALOGV("out_get_sample_rate() returns %u", out_rate);
return out_rate;
}
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->config.output_sample_rate) {
+ ALOGE("out_set_sample_rate(rate=%u) resampling enabled can't change sample rate from "
+ "%u to %u", out->dev->config.output_sample_rate, rate);
+ return -ENOSYS;
+ }
+#endif // ENABLE_RESAMPLING
if (!sample_rate_supported(rate)) {
ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate);
return -ENOSYS;
}
- struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream);
SUBMIX_ALOGV("out_set_sample_rate(rate=%u)", rate);
out->dev->config.common.sample_rate = rate;
return 0;
@@ -628,7 +669,13 @@
const struct submix_config * const config = &out->dev->config;
const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
&stream->common, config, config->buffer_size_frames);
+#if ENABLE_RESAMPLING
+ // Sample rate conversion occurs when data is read from the input so data in the buffer is
+ // at output_sample_rate Hz.
+ const uint32_t latency_ms = (buffer_size_frames * 1000) / config->output_sample_rate;
+#else
const uint32_t latency_ms = (buffer_size_frames * 1000) / config->common.sample_rate;
+#endif // ENABLE_RESAMPLING
SUBMIX_ALOGV("out_get_latency() returns %u ms, size in frames %zu, sample rate %u",
latency_ms, buffer_size_frames, config->common.sample_rate);
return latency_ms;
@@ -728,7 +775,7 @@
return 0;
}
const ssize_t written_bytes = written_frames * frame_size;
- SUBMIX_ALOGV("out_write() wrote %zd bytes %zd frames)", written_bytes, written_frames);
+ SUBMIX_ALOGV("out_write() wrote %zd bytes %zd frames", written_bytes, written_frames);
return written_bytes;
}
@@ -767,13 +814,27 @@
{
const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
const_cast<struct audio_stream*>(stream));
- SUBMIX_ALOGV("in_get_sample_rate() returns %u", in->dev->config.common.sample_rate);
- return in->dev->config.common.sample_rate;
+#if ENABLE_RESAMPLING
+ const uint32_t rate = in->dev->config.input_sample_rate;
+#else
+ const uint32_t rate = in->dev->config.common.sample_rate;
+#endif // ENABLE_RESAMPLING
+ SUBMIX_ALOGV("in_get_sample_rate() returns %u", rate);
+ return rate;
}
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->config.input_sample_rate) {
+ ALOGE("in_set_sample_rate(rate=%u) resampling enabled can't change sample rate from "
+ "%u to %u", in->dev->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;
@@ -788,8 +849,15 @@
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 buffer_size_frames = calculate_stream_pipe_size_in_frames(
+ size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
stream, config, config->buffer_period_size_frames);
+#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 * audio_stream_frame_size(stream);
SUBMIX_ALOGV("in_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes,
buffer_size_frames);
@@ -871,7 +939,6 @@
static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
size_t bytes)
{
- ssize_t frames_read = -1977;
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;
@@ -929,8 +996,37 @@
}
#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->config.output_sample_rate;
+ const size_t resampler_buffer_size_frames =
+ sizeof(rsxadev->resampler_buffer) / sizeof(rsxadev->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->config.common.format == AUDIO_FORMAT_PCM_16_BIT);
+ ALOG_ASSERT(get_channel_count_from_mask(rsxadev->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->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
@@ -973,6 +1069,28 @@
}
#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);