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Commit ebbbb7a2 authored by Jakub Pawłowski's avatar Jakub Pawłowski Committed by Automerger Merge Worker
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Merge "LE Audio: accept 24 bit audio from audio framework properly" am: 2a9456ee 

Original change: https://android-review.googlesource.com/c/platform/packages/modules/Bluetooth/+/2087344



Change-Id: I417da39c1e0f4ba6473a5f78e859f8c49f5f5d38
Signed-off-by: default avatarAutomerger Merge Worker <android-build-automerger-merge-worker@system.gserviceaccount.com>
parents f87a0a2c 2a9456ee

system/bta/le_audio/client.cc

+78 −46
Original line number Diff line number Diff line
@@ -116,6 +116,23 @@ std::ostream& operator<<(std::ostream& os, const AudioState& audio_state) { 
namespace {

void le_audio_gattc_callback(tBTA_GATTC_EVT event, tBTA_GATTC* p_data);



inline uint8_t bits_to_bytes_per_sample(uint8_t bits_per_sample) {

  // 24 bit audio stream is sent as unpacked, each sample takes 4 bytes.

  if (bits_per_sample == 24) return 4;



  return bits_per_sample / 8;

}



inline lc3_pcm_format bits_to_lc3_bits(uint8_t bits_per_sample) {

  if (bits_per_sample == 16) return LC3_PCM_FORMAT_S16;



  if (bits_per_sample == 24) return LC3_PCM_FORMAT_S24;



  LOG_ALWAYS_FATAL("Encoder/decoder don't know how to handle %d",

                   bits_per_sample);

  return LC3_PCM_FORMAT_S16;

}



class LeAudioClientImpl;

LeAudioClientImpl* instance;

LeAudioClientAudioSinkReceiver* audioSinkReceiver;
@@ -1923,27 +1940,36 @@ class LeAudioClientImpl : public LeAudioClient { 
    return true;

  }



  void get_mono_stream(const std::vector<uint8_t>& data,

                       std::vector<int16_t>& chan_mono, int pitch = 1) {

    uint16_t num_of_frames_per_ch;



    int dt_us = current_source_codec_config.data_interval_us;

    int af_hz = audio_framework_source_config.sample_rate;

    num_of_frames_per_ch = lc3_frame_samples(dt_us, af_hz);



    chan_mono.reserve(num_of_frames_per_ch);

    for (int i = 0; i < pitch * num_of_frames_per_ch; i += pitch) {

      const uint8_t* sample = data.data() + i * 4;



      int16_t left = (int16_t)((*(sample + 1) << 8) + *sample) >> 1;



      sample += 2;

      int16_t right = (int16_t)((*(sample + 1) << 8) + *sample) >> 1;



      uint16_t mono_data = (int16_t)(((uint32_t)left + (uint32_t)right) >> 1);



      chan_mono.push_back(mono_data);

  // mix stero signal into mono

  std::vector<uint8_t> mono_blend(const std::vector<uint8_t>& buf,

                                  int bytes_per_sample, size_t frames) {

    std::vector<uint8_t> mono_out;

    mono_out.resize(frames * bytes_per_sample);



    if (bytes_per_sample == 2) {

      int16_t* out = (int16_t*)mono_out.data();

      for (size_t i = 0; i < frames; ++i) {

        const int16_t* in = (int16_t*)(buf.data());

        int accum = 0;

        accum += *in++;

        accum += *in++;

        accum /= 2;  // round to 0

        *out++ = accum;

      }

    } else if (bytes_per_sample == 4) {

      int32_t* out = (int32_t*)mono_out.data();

      for (size_t i = 0; i < frames; ++i) {

        const int32_t* in = (int32_t*)(buf.data());

        int accum = 0;

        accum += *in++;

        accum += *in++;

        accum /= 2;  // round to 0

        *out++ = accum;

      }

    } else {

      LOG_ERROR("Don't know how to mono blend that %d!", bytes_per_sample);

    }

    return mono_out;

  }



  void PrepareAndSendToTwoDevices(
@@ -1958,6 +1984,11 @@ class LeAudioClientImpl : public LeAudioClient { 
    int af_hz = audio_framework_source_config.sample_rate;

    number_of_required_samples_per_channel = lc3_frame_samples(dt_us, af_hz);



    lc3_pcm_format bits_per_sample =

        bits_to_lc3_bits(audio_framework_source_config.bits_per_sample);

    uint8_t bytes_per_sample =

        bits_to_bytes_per_sample(audio_framework_source_config.bits_per_sample);



    for (auto [cis_handle, audio_location] : stream_conf->sink_streams) {

      if (audio_location & le_audio::codec_spec_conf::kLeAudioLocationAnyLeft)

        left_cis_handle = cis_handle;
@@ -1965,11 +1996,12 @@ class LeAudioClientImpl : public LeAudioClient { 
        right_cis_handle = cis_handle;

    }



    if (data.size() < 2 /* bytes per sample */ * 2 /* channels */ *

    if (data.size() < bytes_per_sample * 2 /* channels */ *

                          number_of_required_samples_per_channel) {

      LOG(ERROR) << __func__ << " Missing samples. Data size: " << +data.size()

                 << " expected: "

                 << 2 * 2 * number_of_required_samples_per_channel;

                 << bytes_per_sample * 2 *

                        number_of_required_samples_per_channel;

      return;

    }
@@ -1979,26 +2011,22 @@ class LeAudioClientImpl : public LeAudioClient { 
    bool mono = (left_cis_handle == 0) || (right_cis_handle == 0);



    if (!mono) {

      lc3_encode(lc3_encoder_left, LC3_PCM_FORMAT_S16,

                 (const int16_t*)data.data(), 2, chan_left_enc.size(),

                 chan_left_enc.data());

      lc3_encode(lc3_encoder_right, LC3_PCM_FORMAT_S16,

                 ((const int16_t*)data.data()) + 1, 2, chan_right_enc.size(),

      lc3_encode(lc3_encoder_left, bits_per_sample, data.data(), 2,

                 chan_left_enc.size(), chan_left_enc.data());

      lc3_encode(lc3_encoder_right, bits_per_sample,

                 data.data() + bytes_per_sample, 2, chan_right_enc.size(),

                 chan_right_enc.data());

    } else {

      std::vector<int16_t> chan_mono;

      get_mono_stream(data, chan_mono);



      std::vector<uint8_t> mono = mono_blend(

          data, bytes_per_sample, number_of_required_samples_per_channel);

      if (left_cis_handle) {

        lc3_encode(lc3_encoder_left, LC3_PCM_FORMAT_S16,

                   (const int16_t*)chan_mono.data(), 1, chan_left_enc.size(),

                   chan_left_enc.data());

        lc3_encode(lc3_encoder_left, bits_per_sample, mono.data(), 1,

                   chan_left_enc.size(), chan_left_enc.data());

      }



      if (right_cis_handle) {

        lc3_encode(lc3_encoder_right, LC3_PCM_FORMAT_S16,

                   (const int16_t*)chan_mono.data(), 1, chan_right_enc.size(),

                   chan_right_enc.data());

        lc3_encode(lc3_encoder_right, bits_per_sample, mono.data(), 1,

                   chan_right_enc.size(), chan_right_enc.data());

      }

    }
@@ -2025,6 +2053,10 @@ class LeAudioClientImpl : public LeAudioClient { 
    int dt_us = current_source_codec_config.data_interval_us;

    int af_hz = audio_framework_source_config.sample_rate;

    number_of_required_samples_per_channel = lc3_frame_samples(dt_us, af_hz);

    lc3_pcm_format bits_per_sample =

        bits_to_lc3_bits(audio_framework_source_config.bits_per_sample);

    uint8_t bytes_per_sample =

        bits_to_bytes_per_sample(audio_framework_source_config.bits_per_sample);



    if ((int)data.size() < (2 /* bytes per sample */ * num_channels *

                            number_of_required_samples_per_channel)) {
@@ -2036,21 +2068,19 @@ class LeAudioClientImpl : public LeAudioClient { 
    if (num_channels == 1) {

      /* Since we always get two channels from framework, lets make it mono here

       */

      std::vector<int16_t> chan_mono;

      get_mono_stream(data, chan_mono);

      std::vector<uint8_t> mono = mono_blend(

          data, bytes_per_sample, number_of_required_samples_per_channel);



      auto err = lc3_encode(lc3_encoder_left, LC3_PCM_FORMAT_S16,

                            (const int16_t*)chan_mono.data(), 1, byte_count,

                            chan_encoded.data());

      auto err = lc3_encode(lc3_encoder_left, bits_per_sample, mono.data(), 1,

                            byte_count, chan_encoded.data());



      if (err < 0) {

        LOG(ERROR) << " error while encoding, error code: " << +err;

      }

    } else {

      lc3_encode(lc3_encoder_left, LC3_PCM_FORMAT_S16,

                 (const int16_t*)data.data(), 2, byte_count,

                 chan_encoded.data());

      lc3_encode(lc3_encoder_right, LC3_PCM_FORMAT_S16,

      lc3_encode(lc3_encoder_left, bits_per_sample, (const int16_t*)data.data(),

                 2, byte_count, chan_encoded.data());

      lc3_encode(lc3_encoder_right, bits_per_sample,

                 (const int16_t*)data.data() + 1, 2, byte_count,

                 chan_encoded.data() + byte_count);

    }
@@ -2250,6 +2280,8 @@ class LeAudioClientImpl : public LeAudioClient { 


    int dt_us = current_sink_codec_config.data_interval_us;

    int af_hz = audio_framework_sink_config.sample_rate;

    lc3_pcm_format bits_per_sample =

        bits_to_lc3_bits(audio_framework_sink_config.bits_per_sample);



    int pcm_size;

    if (dt_us == 10000) {
@@ -2282,7 +2314,7 @@ class LeAudioClientImpl : public LeAudioClient { 
    lc3_decoder_t decoder_to_use =

        is_left ? lc3_decoder_left : lc3_decoder_right;



    err = lc3_decode(decoder_to_use, data, size, LC3_PCM_FORMAT_S16,

    err = lc3_decode(decoder_to_use, data, size, bits_per_sample,

                     pcm_data_decoded.data(), 1 /* pitch */);



    if (err < 0) {