Hi Andrzej,
On Wed, Jan 15, 2014 at 11:59 AM, Andrzej Kaczmarek
<andrzej.kaczmarek@xxxxxxxxx> wrote:
> This patch adds support to register audio enpoints via Audio IPC.
> Endpoints are registered based on predefined codecs table and for
> each defined codec one endpoint is registered. By default, only
> SBC will be supported.
> ---
> android/hal-audio.c | 167 ++++++++++++++++++++++++++++++++++++++++++++++++++++
> 1 file changed, 167 insertions(+)
>
> diff --git a/android/hal-audio.c b/android/hal-audio.c
> index 354c3cf..84e7348 100644
> --- a/android/hal-audio.c
> +++ b/android/hal-audio.c
> @@ -31,6 +31,11 @@
> #include "audio-msg.h"
> #include "hal-log.h"
> #include "hal-msg.h"
> +#include "../profiles/audio/a2dp-codecs.h"
> +
> +static const uint8_t a2dp_src_uuid[] = {
> + 0x00, 0x00, 0x11, 0x0a, 0x00, 0x00, 0x10, 0x00,
> + 0x80, 0x00, 0x00, 0x80, 0x5f, 0x9b, 0x34, 0xfb };
>
> static int listen_sk = -1;
> static int audio_sk = -1;
> @@ -40,11 +45,118 @@ static pthread_t ipc_th = 0;
> static pthread_mutex_t close_mutex = PTHREAD_MUTEX_INITIALIZER;
> static pthread_mutex_t sk_mutex = PTHREAD_MUTEX_INITIALIZER;
>
> +struct audio_input_config {
> + uint32_t rate;
> + uint32_t channels;
> + audio_format_t format;
> +};
> +
> +static int codec_sbc_get_presets(struct audio_preset *preset, size_t *len);
> +
> +struct audio_codec {
> + uint8_t type;
> +
> + int (*get_presets) (struct audio_preset *preset, size_t *len);
> +
> + int (*init) (struct audio_preset *preset, void **codec_data);
> + int (*cleanup) (void *codec_data);
> + int (*get_config) (void *codec_data,
> + struct audio_input_config *config);
> + ssize_t (*write_data) (void *codec_data, const void *buffer,
> + size_t bytes);
> +};
> +
> +static const struct audio_codec audio_codecs[] = {
> + {
> + .type = A2DP_CODEC_SBC,
> +
> + .get_presets = codec_sbc_get_presets,
> + }
> +};
> +
> +#define NUM_CODECS (sizeof(audio_codecs) / sizeof(audio_codecs[0]))
> +
> +#define MAX_AUDIO_ENDPOINTS NUM_CODECS
> +
> +struct audio_endpoint {
> + uint8_t id;
> + const struct audio_codec *codec;
> + void *codec_data;
> + int fd;
> +};
> +
> +static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
> +
> struct a2dp_audio_dev {
> struct audio_hw_device dev;
> struct audio_stream_out *out;
> };
>
> +static const a2dp_sbc_t codec_sbc_presets[] = {
Call it sbc_presets
> + {
> + .frequency = SBC_SAMPLING_FREQ_44100 | SBC_SAMPLING_FREQ_48000,
> + .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,
> + .allocation_method = SBC_ALLOCATION_SNR |
> + SBC_ALLOCATION_LOUDNESS,
> + .block_length = SBC_BLOCK_LENGTH_4 | SBC_BLOCK_LENGTH_8 |
> + SBC_BLOCK_LENGTH_12 | SBC_BLOCK_LENGTH_16,
> + .min_bitpool = MIN_BITPOOL,
> + .max_bitpool = MAX_BITPOOL
> + },
> + {
> + .frequency = SBC_SAMPLING_FREQ_44100,
> + .channel_mode = SBC_CHANNEL_MODE_STEREO,
> + .subbands = SBC_SUBBANDS_8,
> + .allocation_method = SBC_ALLOCATION_LOUDNESS,
> + .block_length = SBC_BLOCK_LENGTH_16,
> + .min_bitpool = MIN_BITPOOL,
> + .max_bitpool = MAX_BITPOOL
> + },
> + {
> + .frequency = SBC_SAMPLING_FREQ_48000,
> + .channel_mode = SBC_CHANNEL_MODE_STEREO,
> + .subbands = SBC_SUBBANDS_8,
> + .allocation_method = SBC_ALLOCATION_LOUDNESS,
> + .block_length = SBC_BLOCK_LENGTH_16,
> + .min_bitpool = MIN_BITPOOL,
> + .max_bitpool = MAX_BITPOOL
> + },
> +};
Perhaps we should add the recomended values from A2DP spec, iirc there
are at least 3 preset recommended there, high, medium, low quality.
> +static int codec_sbc_get_presets(struct audio_preset *preset, size_t *len)
> +{
Call this one sbc_get_presets
> + int i;
> + int count;
> + size_t new_len = 0;
> + uint8_t *ptr = (uint8_t *) preset;
> + size_t preset_size = sizeof(*preset) + sizeof(a2dp_sbc_t);
> +
> + DBG("");
> +
> + count = sizeof(codec_sbc_presets) / sizeof(codec_sbc_presets[0]);
> +
> + for (i = 0; i < count; i++) {
> + preset = (struct audio_preset *) ptr;
> +
> + if (new_len + preset_size > *len)
> + break;
> +
> + preset->len = sizeof(a2dp_sbc_t);
> + memcpy(preset->data, &codec_sbc_presets[i], preset->len);
> +
> + new_len += preset_size;
> + ptr += preset_size;
> + }
> +
> + *len = new_len;
> +
> + return i;
> +}
> +
> static void audio_ipc_cleanup(void)
> {
> if (audio_sk >= 0) {
> @@ -200,6 +312,54 @@ failed:
> return AUDIO_STATUS_FAILED;
> }
>
> +static int ipc_open_cmd(const struct audio_codec *codec)
> +{
> + uint8_t buf[BLUEZ_AUDIO_MTU];
> + struct audio_cmd_open *cmd = (struct audio_cmd_open *) buf;
> + struct audio_rsp_open rsp;
> + size_t cmd_len = sizeof(buf) - sizeof(*cmd);
> + size_t rsp_len = sizeof(rsp);
> + int result;
> +
> + DBG("");
> +
> + memcpy(cmd->uuid, a2dp_src_uuid, sizeof(a2dp_src_uuid));
> +
> + cmd->codec = codec->type;
> + cmd->presets = codec->get_presets(cmd->preset, &cmd_len);
> +
> + cmd_len += sizeof(*cmd);
> +
> + result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN, cmd_len, cmd,
> + &rsp_len, &rsp, NULL);
> +
> + if (result != AUDIO_STATUS_SUCCESS)
> + return 0;
> +
> + return rsp.id;
> +}
> +
> +static int register_endpoints(void)
> +{
> + struct audio_endpoint *ep = &audio_endpoints[0];
> + size_t i;
> +
> + for (i = 0; i < NUM_CODECS; i++) {
> + const struct audio_codec *codec = &audio_codecs[i];
> +
> + ep->id = ipc_open_cmd(codec);
> +
> + if (!ep->id)
> + return AUDIO_STATUS_FAILED;
> +
> + ep->codec = codec;
> + ep->codec_data = NULL;
> + ep->fd = -1;
> + }
This seems to be overwriting ep if there is more than one codec
available, I guess you need to move ep within the loop as well then
you assign the correspondent codec index to the endpoint index.
--
Luiz Augusto von Dentz
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