Hi Andrzej,
On Wed, Apr 9, 2014 at 5:16 PM, Andrzej Kaczmarek
<andrzej.kaczmarek@xxxxxxxxx> wrote:
> This patch adds new codec abstraction call which can be used to adjust
> audio quality while playing. As for now we can either decrease quality
> or restore default one.
>
> It's up to actual codec capabilities and implementation how this can be
> handled. In case of SBC bitpool is decreased by fixed amount (5) until
> min allowable value is reached (33) - the same values are used in
> PulseAudio.
> ---
> android/hal-audio.c | 81 ++++++++++++++++++++++++++++++++++++++++++-----------
> 1 file changed, 65 insertions(+), 16 deletions(-)
>
> diff --git a/android/hal-audio.c b/android/hal-audio.c
> index e58be2b..2db927c 100644
> --- a/android/hal-audio.c
> +++ b/android/hal-audio.c
> @@ -47,6 +47,9 @@
>
> #define MAX_DELAY 100000 /* 100ms */
>
> +#define SBC_QUALITY_MIN_BITPOOL 33
> +#define SBC_QUALITY_STEP 5
> +
> static const uint8_t a2dp_src_uuid[] = {
> 0x00, 0x00, 0x11, 0x0a, 0x00, 0x00, 0x10, 0x00,
> 0x80, 0x00, 0x00, 0x80, 0x5f, 0x9b, 0x34, 0xfb };
> @@ -128,6 +131,8 @@ struct sbc_data {
>
> sbc_t enc;
>
> + uint16_t payload_len;
> +
> size_t in_frame_len;
> size_t in_buf_size;
>
> @@ -189,6 +194,10 @@ static size_t sbc_get_mediapacket_duration(void *codec_data);
> static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
> size_t len, struct media_packet *mp,
> size_t mp_data_len, size_t *written);
> +static bool sbc_quality_ctrl(void *codec_data, uint8_t op);
> +
> +#define QUALITY_CTRL_DEFAULT 0x00
> +#define QUALITY_CTRL_DECREASE 0x01
Lets call it QOS_POLICY_*
> struct audio_codec {
> uint8_t type;
> @@ -205,6 +214,7 @@ struct audio_codec {
> ssize_t (*encode_mediapacket) (void *codec_data, const uint8_t *buffer,
> size_t len, struct media_packet *mp,
> size_t mp_data_len, size_t *written);
> + bool (*quality_ctrl) (void *codec_data, uint8_t op);
I prefer to use update_qos
> };
>
> static const struct audio_codec audio_codecs[] = {
> @@ -219,6 +229,7 @@ static const struct audio_codec audio_codecs[] = {
> .get_buffer_size = sbc_get_buffer_size,
> .get_mediapacket_duration = sbc_get_mediapacket_duration,
> .encode_mediapacket = sbc_encode_mediapacket,
> + .quality_ctrl = sbc_quality_ctrl,
sbc_update_qos
> }
> };
>
> @@ -412,14 +423,33 @@ static void sbc_init_encoder(struct sbc_data *sbc_data)
> in->min_bitpool, in->max_bitpool);
> }
>
> -static int sbc_codec_init(struct audio_preset *preset, uint16_t payload_len,
> - void **codec_data)
> +static void sbc_codec_calculate(struct sbc_data *sbc_data)
> {
> - struct sbc_data *sbc_data;
> size_t in_frame_len;
> size_t out_frame_len;
> size_t num_frames;
>
> + in_frame_len = sbc_get_codesize(&sbc_data->enc);
> + out_frame_len = sbc_get_frame_length(&sbc_data->enc);
> + num_frames = sbc_data->payload_len / out_frame_len;
> +
> + sbc_data->in_frame_len = in_frame_len;
> + sbc_data->in_buf_size = num_frames * in_frame_len;
> +
> + sbc_data->out_frame_len = out_frame_len;
> +
> + sbc_data->frame_duration = sbc_get_frame_duration(&sbc_data->enc);
> + sbc_data->frames_per_packet = num_frames;
> +
> + DBG("in_frame_len=%zu out_frame_len=%zu frames_per_packet=%zu",
> + in_frame_len, out_frame_len, num_frames);
> +}
Looks like you remembered to update the frame size, however how about
the latency? Does it gets updated automagically?
> +static int sbc_codec_init(struct audio_preset *preset, uint16_t payload_len,
> + void **codec_data)
> +{
> + struct sbc_data *sbc_data;
> +
> if (preset->len != sizeof(a2dp_sbc_t)) {
> error("SBC: preset size mismatch");
> return AUDIO_STATUS_FAILED;
> @@ -433,20 +463,9 @@ static int sbc_codec_init(struct audio_preset *preset, uint16_t payload_len,
>
> sbc_init_encoder(sbc_data);
>
> - in_frame_len = sbc_get_codesize(&sbc_data->enc);
> - out_frame_len = sbc_get_frame_length(&sbc_data->enc);
> - num_frames = payload_len / out_frame_len;
> -
> - sbc_data->in_frame_len = in_frame_len;
> - sbc_data->in_buf_size = num_frames * in_frame_len;
> -
> - sbc_data->out_frame_len = out_frame_len;
> -
> - sbc_data->frame_duration = sbc_get_frame_duration(&sbc_data->enc);
> - sbc_data->frames_per_packet = num_frames;
> + sbc_data->payload_len = payload_len;
>
> - DBG("in_frame_len=%zu out_frame_len=%zu frames_per_packet=%zu",
> - in_frame_len, out_frame_len, num_frames);
> + sbc_codec_calculate(sbc_data);
>
> *codec_data = sbc_data;
>
> @@ -541,6 +560,36 @@ static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
> return consumed;
> }
>
> +static bool sbc_quality_ctrl(void *codec_data, uint8_t op)
> +{
> + struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
> + uint8_t curr_bitpool = sbc_data->enc.bitpool;
> + uint8_t new_bitpool = curr_bitpool;
> +
> + switch (op) {
> + case QUALITY_CTRL_DEFAULT:
> + new_bitpool = sbc_data->sbc.max_bitpool;
> + break;
> +
> + case QUALITY_CTRL_DECREASE:
> + new_bitpool = curr_bitpool - SBC_QUALITY_STEP;
> + if (new_bitpool < SBC_QUALITY_MIN_BITPOOL)
> + new_bitpool = SBC_QUALITY_MIN_BITPOOL;
> + break;
> + }
> +
> + if (new_bitpool == curr_bitpool)
> + return false;
> +
> + sbc_data->enc.bitpool = new_bitpool;
> +
> + sbc_codec_calculate(sbc_data);
> +
> + info("SBC: bitpool chaged: %d -> %d", curr_bitpool, new_bitpool);
> +
> + return true;
> +}
> +
> static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
> void *param, size_t *rsp_len, void *rsp, int *fd)
> {
> --
> 1.9.1
>
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--
Luiz Augusto von Dentz
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