From: Andrei Emeltchenko <andrei.emeltchenko@xxxxxxxxx>
Resample Android audio from 44100 to 8000.
---
android/hal-sco.c | 112 ++++++++++++++++++++++++++++++++++++++++++++++++++----
1 file changed, 105 insertions(+), 7 deletions(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index f74c1c7..7b6608b 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -27,6 +27,7 @@
#include <hardware/audio.h>
#include <hardware/hardware.h>
+#include <audio_utils/resampler.h>
#include "../src/shared/util.h"
#include "sco-msg.h"
@@ -34,6 +35,7 @@
#include "hal-log.h"
#define AUDIO_STREAM_DEFAULT_RATE 44100
+#define AUDIO_STREAM_SCO_RATE 8000
#define AUDIO_STREAM_DEFAULT_FORMAT AUDIO_FORMAT_PCM_16_BIT
#define OUT_BUFFER_SIZE 2560
@@ -60,6 +62,10 @@ struct sco_stream_out {
int fd;
uint8_t *downmix_buf;
+
+ struct resampler_itfe *resampler;
+ int16_t *resample_buf;
+ uint32_t resample_frame_num;
};
struct sco_audio_dev {
@@ -67,6 +73,22 @@ struct sco_audio_dev {
struct sco_stream_out *out;
};
+/*
+ * return the minimum frame numbers from resampling between BT stack's rate
+ * and audio flinger's. For output stream, 'output' shall be true, otherwise
+ * false for input streams at audio flinger side.
+ */
+static size_t get_resample_frame_num(uint32_t sco_rate, uint32_t rate,
+ size_t frame_num, bool output)
+{
+ size_t resample_frames_num = frame_num * sco_rate / rate + output;
+
+ DBG("resampler: sco_rate %d frame_num %zd rate %d resample frames %zd",
+ sco_rate, frame_num, rate, resample_frames_num);
+
+ return resample_frames_num;
+}
+
/* Audio IPC functions */
static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
@@ -258,6 +280,9 @@ static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
{
struct sco_stream_out *out = (struct sco_stream_out *) stream;
size_t frame_num = bytes / audio_stream_frame_size(&out->stream.common);
+ size_t output_frame_num = frame_num;
+ void *send_buf = out->downmix_buf;
+ size_t total;
DBG("write to fd %d bytes %zu", out->fd, bytes);
@@ -267,6 +292,34 @@ static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
}
downmix_to_mono(out, buffer, frame_num);
+
+ if (out->resampler) {
+ int ret;
+
+ /* limit resampler's output within what resample buf can hold */
+ output_frame_num = out->resample_frame_num;
+
+ ret = out->resampler->resample_from_input(out->resampler,
+ send_buf,
+ &frame_num,
+ out->resample_buf,
+ &output_frame_num);
+ if (ret) {
+ error("Failed to resample frames: %zd input %zd (%s)",
+ frame_num, output_frame_num, strerror(ret));
+ return -1;
+ }
+
+ send_buf = out->resample_buf;
+
+ DBG("Resampled: frame_num %zd, output_frame_num %zd",
+ frame_num, output_frame_num);
+ }
+
+ total = output_frame_num * sizeof(int16_t) * 1;
+
+ DBG("total %zd", total);
+
return bytes;
}
@@ -299,19 +352,18 @@ static size_t out_get_buffer_size(const struct audio_stream *stream)
static uint32_t out_get_channels(const struct audio_stream *stream)
{
- DBG("");
+ struct sco_stream_out *out = (struct sco_stream_out *) stream;
- /* AudioFlinger can only provide stereo stream, so we return it here and
- * later we'll downmix this to mono in case codec requires it
- */
- return AUDIO_CHANNEL_OUT_STEREO;
+ DBG("channels num: %u", popcount(out->cfg.channels));
+
+ return out->cfg.channels;
}
static audio_format_t out_get_format(const struct audio_stream *stream)
{
struct sco_stream_out *out = (struct sco_stream_out *) stream;
- DBG("");
+ DBG("format: %u", out->cfg.format);
return out->cfg.format;
}
@@ -401,6 +453,8 @@ static int audio_open_output_stream(struct audio_hw_device *dev,
struct sco_audio_dev *adev = (struct sco_audio_dev *) dev;
struct sco_stream_out *out;
int fd = -1;
+ int chan_num, ret;
+ size_t resample_size;
uint16_t mtu;
DBG("");
@@ -433,8 +487,9 @@ static int audio_open_output_stream(struct audio_hw_device *dev,
out->stream.write = out_write;
out->stream.get_render_position = out_get_render_position;
+ /* Configuration for Android */
out->cfg.format = AUDIO_STREAM_DEFAULT_FORMAT;
- out->cfg.channels = AUDIO_CHANNEL_OUT_MONO;
+ out->cfg.channels = AUDIO_CHANNEL_OUT_STEREO;
out->cfg.rate = AUDIO_STREAM_DEFAULT_RATE;
out->cfg.frame_num = OUT_STREAM_FRAMES;
out->cfg.mtu = mtu;
@@ -446,11 +501,54 @@ static int audio_open_output_stream(struct audio_hw_device *dev,
}
DBG("size %zd", out_get_buffer_size(&out->stream.common));
+
+ /* Channel numbers for resampler */
+ chan_num = 1;
+
+ ret = create_resampler(out->cfg.rate, AUDIO_STREAM_SCO_RATE, chan_num,
+ RESAMPLER_QUALITY_DEFAULT, NULL,
+ &out->resampler);
+ if (ret) {
+ error("Failed to create resampler (%s)", strerror(ret));
+ goto failed;
+ }
+
+ DBG("Created resampler: input rate [%d] output rate [%d] channels [%d]",
+ out->cfg.rate, AUDIO_STREAM_SCO_RATE, chan_num);
+
+ out->resample_frame_num = get_resample_frame_num(AUDIO_STREAM_SCO_RATE,
+ out->cfg.rate,
+ out->cfg.frame_num, 1);
+
+ if (!out->resample_frame_num) {
+ error("frame num is too small to resample, discard it");
+ goto failed;
+ }
+
+ resample_size = sizeof(int16_t) * chan_num * out->resample_frame_num;
+
+ out->resample_buf = malloc(resample_size);
+ if (!out->resample_buf) {
+ error("failed to allocate resample buffer for %u frames",
+ out->resample_frame_num);
+ goto failed;
+ }
+
+ DBG("resampler: frame num %u buf size %zd bytes",
+ out->resample_frame_num, resample_size);
+
*stream_out = &out->stream;
adev->out = out;
out->fd = fd;
return 0;
+failed:
+ free(out->downmix_buf);
+ free(out);
+ stream_out = NULL;
+ adev->out = NULL;
+
+ return ret;
}
static void audio_close_output_stream(struct audio_hw_device *dev,
--
1.8.3.2
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