This patch adds simple downmix support from stereo to mono in order to
support mono channel mode as it's mandatory for SBC codec. It uses
simple (L+R)/2 calculation which should be good enough.
---
android/hal-audio.c | 54 ++++++++++++++++++++++++++++++++++++++++++++++++-----
1 file changed, 49 insertions(+), 5 deletions(-)
diff --git a/android/hal-audio.c b/android/hal-audio.c
index efdf823..f90265b 100644
--- a/android/hal-audio.c
+++ b/android/hal-audio.c
@@ -36,9 +36,12 @@
#include "hal-log.h"
#include "hal-msg.h"
#include "../profiles/audio/a2dp-codecs.h"
+#include "../src/shared/util.h"
#define FIXED_A2DP_PLAYBACK_LATENCY_MS 25
+#define FIXED_BUFFER_SIZE (20 * 512)
+
#define MAX_FRAMES_IN_PAYLOAD 15
static const uint8_t a2dp_src_uuid[] = {
@@ -220,6 +223,8 @@ struct a2dp_stream_out {
struct audio_endpoint *ep;
enum a2dp_state_t audio_state;
struct audio_input_config cfg;
+
+ uint8_t *downmix_buf;
};
struct a2dp_audio_dev {
@@ -230,7 +235,8 @@ struct a2dp_audio_dev {
static const a2dp_sbc_t sbc_presets[] = {
{
.frequency = SBC_SAMPLING_FREQ_44100 | SBC_SAMPLING_FREQ_48000,
- .channel_mode = SBC_CHANNEL_MODE_DUAL_CHANNEL |
+ .channel_mode = SBC_CHANNEL_MODE_MONO |
+ SBC_CHANNEL_MODE_DUAL_CHANNEL |
SBC_CHANNEL_MODE_STEREO |
SBC_CHANNEL_MODE_JOINT_STEREO,
.subbands = SBC_SUBBANDS_4 | SBC_SUBBANDS_8,
@@ -826,6 +832,21 @@ static void unregister_endpoints(void)
}
}
+static void downmix_to_mono(struct a2dp_stream_out *out, const uint8_t *buffer,
+ size_t bytes)
+{
+ const int16_t *input = (const void *) buffer;
+ int16_t *output = (void *) out->downmix_buf;
+ size_t i;
+
+ for (i = 0; i < bytes / 2; i++) {
+ int16_t l = le16_to_cpu(get_unaligned(&input[i * 2]));
+ int16_t r = le16_to_cpu(get_unaligned(&input[i * 2 + 1]));
+
+ put_unaligned(cpu_to_le16((l + r) / 2), &output[i]);
+ }
+}
+
static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
size_t bytes)
{
@@ -853,6 +874,18 @@ static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
return -1;
}
+ if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
+ if (!out->downmix_buf) {
+ error("audio: downmix buffer not initialized");
+ return -1;
+ }
+
+ downmix_to_mono(out, buffer, bytes);
+
+ return out->ep->codec->write_data(out->ep->codec_data,
+ out->downmix_buf, bytes / 2, out->ep->fd) * 2;
+ }
+
return out->ep->codec->write_data(out->ep->codec_data, buffer,
bytes, out->ep->fd);
}
@@ -890,16 +923,18 @@ static size_t out_get_buffer_size(const struct audio_stream *stream)
* use magic value here and out_write code takes care of splitting
* input buffer into multiple media packets.
*/
- return 20 * 512;
+ return FIXED_BUFFER_SIZE;
}
static uint32_t out_get_channels(const struct audio_stream *stream)
{
- struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
-
DBG("");
- return out->cfg.channels;
+ /* 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;
}
static audio_format_t out_get_format(const struct audio_stream *stream)
@@ -1212,6 +1247,12 @@ static int audio_open_output_stream(struct audio_hw_device *dev,
free(preset);
+ if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
+ out->downmix_buf = malloc(FIXED_BUFFER_SIZE / 2);
+ if (!out->downmix_buf)
+ goto fail;
+ }
+
*stream_out = &out->stream;
a2dp_dev->out = out;
@@ -1230,6 +1271,7 @@ static void audio_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream)
{
struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
+ struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
struct audio_endpoint *ep = a2dp_dev->out->ep;
DBG("");
@@ -1243,6 +1285,8 @@ static void audio_close_output_stream(struct audio_hw_device *dev,
ep->codec->cleanup(ep->codec_data);
ep->codec_data = NULL;
+ free(out->downmix_buf);
+
free(stream);
a2dp_dev->out = NULL;
}
--
1.8.5.3
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