Hi Andrzej,
On Tuesday 04 of February 2014 14:38:45 Andrzej Kaczmarek wrote:
> 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;
> }
>
This patch has been applied, thanks.
--
Best regards,
Szymon Janc
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