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Commit c7ed52fc authored by Pavlin Radoslavov's avatar Pavlin Radoslavov
Browse files

Compute the audio data buffer size based on stream characteristics 

Use the audio data stream characteristics (sample rate, bits per sample,
channel mode) to compute the audio stream output buffer size from
the Audio A2DP HAL to the Bluetooth stack.

Previously, the buffer size was pre-computed based on the original
(44.1kHz, 16 bits per sample, Stereo) PCM audio stream used by SBC.
NOTE: We cannot change the buffer size on the receiver side (Bluetooth),
because SO_RCVBUF option does not have effect for Domain Sockets.

Also, removed uipc_linux.h header file, because it is not used.

Test: A2DP streaming and switching the codecs, sample rate, bits per sample
Bug: 35849921
Change-Id: Ib6772f7564442ac18a02876a5aa5e3208e5aade1
(cherry picked from commit b32704f424e0ab4cd33ad670173997f8e3a0a31c)
parent 4de795fa

system/audio_a2dp_hw/audio_a2dp_hw.cc

+114 −4
Original line number Original line Diff line number Diff line
@@ -150,6 +150,10 @@ struct a2dp_stream_in { 
 *****************************************************************************/

 *****************************************************************************/





static size_t out_get_buffer_size(const struct audio_stream* stream);

static size_t out_get_buffer_size(const struct audio_stream* stream);

static size_t audio_stream_compute_buffer_size(

    btav_a2dp_codec_sample_rate_t codec_sample_rate,

    btav_a2dp_codec_bits_per_sample_t codec_bits_per_sample,

    btav_a2dp_codec_channel_mode_t codec_channel_mode);





/*****************************************************************************

/*****************************************************************************

 *  Externs

 *  Externs
@@ -607,6 +611,9 @@ static int a2dp_read_output_audio_config( 
    common->cfg.rate = stream_config.rate;

    common->cfg.rate = stream_config.rate;

    common->cfg.channel_mask = stream_config.channel_mask;

    common->cfg.channel_mask = stream_config.channel_mask;

    common->cfg.format = stream_config.format;

    common->cfg.format = stream_config.format;

    common->buffer_sz = audio_stream_compute_buffer_size(

        codec_config->sample_rate, codec_config->bits_per_sample,

        codec_config->channel_mode);

  }

  }





  INFO(

  INFO(
@@ -712,8 +719,8 @@ static void a2dp_open_ctrl_path(struct a2dp_stream_common* common) { 
  /* retry logic to catch any timing variations on control channel */

  /* retry logic to catch any timing variations on control channel */

  for (i = 0; i < CTRL_CHAN_RETRY_COUNT; i++) {

  for (i = 0; i < CTRL_CHAN_RETRY_COUNT; i++) {

    /* connect control channel if not already connected */

    /* connect control channel if not already connected */

    if ((common->ctrl_fd = skt_connect(A2DP_CTRL_PATH, common->buffer_sz)) >

    if ((common->ctrl_fd = skt_connect(

        0) {

             A2DP_CTRL_PATH, AUDIO_STREAM_CONTROL_OUTPUT_BUFFER_SZ)) > 0) {

      /* success, now check if stack is ready */

      /* success, now check if stack is ready */

      if (check_a2dp_ready(common) == 0) break;

      if (check_a2dp_ready(common) == 0) break;
@@ -927,6 +934,106 @@ static size_t out_get_buffer_size(const struct audio_stream* stream) { 
  return period_size;

  return period_size;

}

}





static size_t audio_stream_compute_buffer_size(

    btav_a2dp_codec_sample_rate_t codec_sample_rate,

    btav_a2dp_codec_bits_per_sample_t codec_bits_per_sample,

    btav_a2dp_codec_channel_mode_t codec_channel_mode) {

  size_t buffer_sz = AUDIO_STREAM_OUTPUT_BUFFER_SZ;  // Default value

  const uint32_t time_period_ms = 20;                // Conservative 20ms

  uint32_t sample_rate;

  uint32_t bits_per_sample;

  uint32_t number_of_channels;



  // Check the codec config sample rate

  switch (codec_sample_rate) {

    case BTAV_A2DP_CODEC_SAMPLE_RATE_44100:

      sample_rate = 44100;

      break;

    case BTAV_A2DP_CODEC_SAMPLE_RATE_48000:

      sample_rate = 48000;

      break;

    case BTAV_A2DP_CODEC_SAMPLE_RATE_88200:

      sample_rate = 88200;

      break;

    case BTAV_A2DP_CODEC_SAMPLE_RATE_96000:

      sample_rate = 96000;

      break;

    case BTAV_A2DP_CODEC_SAMPLE_RATE_176400:

      sample_rate = 176400;

      break;

    case BTAV_A2DP_CODEC_SAMPLE_RATE_192000:

      sample_rate = 192000;

      break;

    case BTAV_A2DP_CODEC_SAMPLE_RATE_NONE:

    default:

      ERROR("Invalid sample rate: 0x%x", codec_sample_rate);

      return buffer_sz;

  }



  // Check the codec config bits per sample

  switch (codec_bits_per_sample) {

    case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_16:

      bits_per_sample = 16;

      break;

    case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_24:

      bits_per_sample = 24;

      break;

    case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_32:

      bits_per_sample = 32;

      break;

    case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_NONE:

    default:

      ERROR("Invalid bits per sample: 0x%x", codec_bits_per_sample);

      return buffer_sz;

  }



  // Check the codec config channel mode

  switch (codec_channel_mode) {

    case BTAV_A2DP_CODEC_CHANNEL_MODE_MONO:

      number_of_channels = 1;

      break;

    case BTAV_A2DP_CODEC_CHANNEL_MODE_STEREO:

      number_of_channels = 2;

      break;

    case BTAV_A2DP_CODEC_CHANNEL_MODE_NONE:

    default:

      ERROR("Invalid channel mode: 0x%x", codec_channel_mode);

      return buffer_sz;

  }



  //

  // The buffer size is computed by using the following formula:

  //

  // AUDIO_STREAM_OUTPUT_BUFFER_SIZE =

  //    (TIME_PERIOD_MS * AUDIO_STREAM_OUTPUT_BUFFER_PERIODS *

  //     SAMPLE_RATE_HZ * NUMBER_OF_CHANNELS * (BITS_PER_SAMPLE / 8)) / 1000

  //

  // AUDIO_STREAM_OUTPUT_BUFFER_PERIODS controls how the socket buffer is

  // divided for AudioFlinger data delivery. The AudioFlinger mixer delivers

  // data in chunks of

  // (AUDIO_STREAM_OUTPUT_BUFFER_SIZE / AUDIO_STREAM_OUTPUT_BUFFER_PERIODS) .

  // If the number of periods is 2, the socket buffer represents "double

  // buffering" of the AudioFlinger mixer buffer.

  //

  // Furthermore, the AudioFlinger expects the buffer size to be a multiple

  // of 16 frames.

  const size_t divisor = (AUDIO_STREAM_OUTPUT_BUFFER_PERIODS * 16 *

                          number_of_channels * bits_per_sample) /

                         8;



  buffer_sz = (time_period_ms * AUDIO_STREAM_OUTPUT_BUFFER_PERIODS *

               sample_rate * number_of_channels * (bits_per_sample / 8)) /

              1000;



  // Adjust the buffer size so it can be divided by the divisor

  const size_t remainder = buffer_sz % divisor;

  if (remainder != 0) {

    buffer_sz += divisor - remainder;

  }



  return buffer_sz;

}



static uint32_t out_get_channels(const struct audio_stream* stream) {

static uint32_t out_get_channels(const struct audio_stream* stream) {

  struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;

  struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;
@@ -1434,8 +1541,11 @@ static int adev_open_output_stream(struct audio_hw_device* dev, 
    config->channel_mask =

    config->channel_mask =

        out_get_channels((const struct audio_stream*)&out->stream);

        out_get_channels((const struct audio_stream*)&out->stream);





    INFO("Output stream config: format=0x%x sample_rate=%d channel_mask=0x%x",

    INFO(

         config->format, config->sample_rate, config->channel_mask);

        "Output stream config: format=0x%x sample_rate=%d channel_mask=0x%x "

        "buffer_sz=%zu",

        config->format, config->sample_rate, config->channel_mask,

        out->common.buffer_sz);

  }

  }

  *stream_out = &out->stream;

  *stream_out = &out->stream;

  a2dp_dev->output = out;

  a2dp_dev->output = out;

system/audio_a2dp_hw/audio_a2dp_hw.h

+1 −0
Original line number Original line Diff line number Diff line
@@ -54,6 +54,7 @@ 
// resulting in mixer sleep and throttling. We increase this to 28 * 512 to help

// resulting in mixer sleep and throttling. We increase this to 28 * 512 to help

// reduce the effect of variable data consumption.

// reduce the effect of variable data consumption.

#define AUDIO_STREAM_OUTPUT_BUFFER_SZ (28 * 512)

#define AUDIO_STREAM_OUTPUT_BUFFER_SZ (28 * 512)

#define AUDIO_STREAM_CONTROL_OUTPUT_BUFFER_SZ 256





// AUDIO_STREAM_OUTPUT_BUFFER_PERIODS controls how the socket buffer is divided

// AUDIO_STREAM_OUTPUT_BUFFER_PERIODS controls how the socket buffer is divided

// for AudioFlinger data delivery. The AudioFlinger mixer delivers data in

// for AudioFlinger data delivery. The AudioFlinger mixer delivers data in

system/udrv/ulinux/uipc_linux.h

deleted100644 → 0
+0 −34
Original line number Original line Diff line number Diff line 
/******************************************************************************

 *

 *  Copyright (C) 2007-2012 Broadcom Corporation

 *

 *  Licensed under the Apache License, Version 2.0 (the "License");

 *  you may not use this file except in compliance with the License.

 *  You may obtain a copy of the License at:

 *

 *  http://www.apache.org/licenses/LICENSE-2.0

 *

 *  Unless required by applicable law or agreed to in writing, software

 *  distributed under the License is distributed on an "AS IS" BASIS,

 *  WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.

 *  See the License for the specific language governing permissions and

 *  limitations under the License.

 *

 ******************************************************************************/

#ifndef _UIPC_LINUX_H_

#define _UIPC_LINUX_H_



typedef int SOCKET;

#define INVALID_SOCKET (SOCKET)(-1)

#define SOCKET_ERROR (-1)



/* tcp/ip socket configuration */

typedef struct {

  char* p_address;

  unsigned int port;

} tUIPC_LINUX_CFG_TCP;



/* Socket configuration for GLGPS interface */

extern tUIPC_LINUX_CFG_TCP uipc_linux_cfg_glgps;



#endif /* _UIPC_LINUX_H_ */