diff options
Diffstat (limited to 'modules/audio_remote_submix/audio_hw.cpp')
| -rw-r--r-- | modules/audio_remote_submix/audio_hw.cpp | 55 |
1 files changed, 39 insertions, 16 deletions
diff --git a/modules/audio_remote_submix/audio_hw.cpp b/modules/audio_remote_submix/audio_hw.cpp index ed3d311..515f150 100644 --- a/modules/audio_remote_submix/audio_hw.cpp +++ b/modules/audio_remote_submix/audio_hw.cpp @@ -159,11 +159,11 @@ typedef struct route_config { // destroyed if both and input and output streams are destroyed. struct submix_stream_out *output; struct submix_stream_in *input; -#if ENABLE_RESAMPLING - // Buffer used as temporary storage for resampled data prior to returning data to the output +#if (ENABLE_RESAMPLING || ENABLE_CHANNEL_CONVERSION) + // Buffer used as temporary storage for audio data prior to returning data to the output // stream. - int16_t resampler_buffer[DEFAULT_PIPE_SIZE_IN_FRAMES]; -#endif // ENABLE_RESAMPLING + int16_t processor_buffer[DEFAULT_PIPE_SIZE_IN_FRAMES]; +#endif } route_config_t; struct submix_audio_device { @@ -474,8 +474,8 @@ static void submix_audio_device_release_pipe_l(struct submix_audio_device * cons rsxadev->routes[route_idx].rsxSource = 0; } memset(rsxadev->routes[route_idx].address, 0, AUDIO_DEVICE_MAX_ADDRESS_LEN); -#ifdef ENABLE_RESAMPLING - memset(rsxadev->routes[route_idx].resampler_buffer, 0, +#if (ENABLE_RESAMPLING || ENABLE_CHANNEL_CONVERSION) + memset(rsxadev->routes[route_idx].processor_buffer, 0, sizeof(int16_t) * DEFAULT_PIPE_SIZE_IN_FRAMES); #endif } @@ -1147,13 +1147,14 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, } #endif // ENABLE_CHANNEL_CONVERSION +#if (ENABLE_RESAMPLING || ENABLE_CHANNEL_CONVERSION) + const size_t processor_buffer_size_frames = + sizeof(rsxadev->routes[in->route_handle].processor_buffer) / frame_size; +#endif #if ENABLE_RESAMPLING const uint32_t input_sample_rate = in_get_sample_rate(&stream->common); const uint32_t output_sample_rate = rsxadev->routes[in->route_handle].config.output_sample_rate; - const size_t resampler_buffer_size_frames = - sizeof(rsxadev->routes[in->route_handle].resampler_buffer) / - sizeof(rsxadev->routes[in->route_handle].resampler_buffer[0]); float resampler_ratio = 1.0f; // Determine whether resampling is required. if (input_sample_rate != output_sample_rate) { @@ -1170,16 +1171,18 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) { ssize_t frames_read = -1977; size_t read_frames = remaining_frames; -#if ENABLE_RESAMPLING +#if (ENABLE_RESAMPLING || ENABLE_CHANNEL_CONVERSION) char* const saved_buff = buff; +#endif +#if ENABLE_RESAMPLING if (resampler_ratio != 1.0f) { // Calculate the number of frames from the pipe that need to be read to generate // the data for the input stream read. const size_t frames_required_for_resampler = (size_t)( (float)read_frames * (float)resampler_ratio); - read_frames = min(frames_required_for_resampler, resampler_buffer_size_frames); + read_frames = min(frames_required_for_resampler, processor_buffer_size_frames); // Read into the resampler buffer. - buff = (char*)rsxadev->routes[in->route_handle].resampler_buffer; + buff = (char*)rsxadev->routes[in->route_handle].processor_buffer; } #endif // ENABLE_RESAMPLING #if ENABLE_CHANNEL_CONVERSION @@ -1187,6 +1190,13 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, // Need to read half the requested frames since the converted output // data will take twice the space (mono->stereo). read_frames /= 2; + } else if (output_channels == 2 && input_channels == 1) { + // If the resampler is active, we already swapped for the processor_buffer + if (resampler_ratio == 1.0f) { + buff = (char*)rsxadev->routes[in->route_handle].processor_buffer; + } + + read_frames = min(read_frames, processor_buffer_size_frames/2); } #endif // ENABLE_CHANNEL_CONVERSION @@ -1205,11 +1215,17 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, if (output_channels == 2 && input_channels == 1) { // Offset into the output stream data in samples. ssize_t output_stream_offset = 0; + // If resampler is active, continue writing to the temporary buffer + int16_t *mixed_buffer = + (resampler_ratio == 1.0f) ? (int16_t*)saved_buff : (int16_t*)buff; for (ssize_t input_stream_frame = 0; input_stream_frame < frames_read; input_stream_frame++, output_stream_offset += 2) { // Average the content from both channels. - data[input_stream_frame] = ((int32_t)data[output_stream_offset] + - (int32_t)data[output_stream_offset + 1]) / 2; + mixed_buffer[input_stream_frame] = ((int32_t)data[output_stream_offset] + + (int32_t)data[output_stream_offset + 1]) / 2; + } + if (resampler_ratio == 1.0f) { + buff = saved_buff; } } else if (output_channels == 1 && input_channels == 2) { // Offset into the input stream data in samples. @@ -1233,13 +1249,20 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, // sampled at a different rate this will result in very nasty aliasing. const float output_stream_frames = (float)frames_read; size_t input_stream_frame = 0; + size_t input_buf_offset = 0, output_buf_offset = 0; for (float output_stream_frame = 0.0f; output_stream_frame < output_stream_frames && input_stream_frame < remaining_frames; output_stream_frame += resampler_ratio, input_stream_frame++) { - resampled_buffer[input_stream_frame] = data[(size_t)output_stream_frame]; + input_buf_offset = input_stream_frame * input_channels; + output_buf_offset = (size_t)output_stream_frame * input_channels; + resampled_buffer[input_buf_offset] = data[output_buf_offset]; + if (input_channels == 2) { + // copy second channel in the frame + resampled_buffer[input_buf_offset + 1] = data[output_buf_offset + 1]; + } } - ALOG_ASSERT(input_stream_frame <= (ssize_t)resampler_buffer_size_frames); + ALOG_ASSERT(input_stream_frame <= (ssize_t)processor_buffer_size_frames); SUBMIX_ALOGV("in_read(): resampler produced %zd frames", input_stream_frame); frames_read = input_stream_frame; buff = saved_buff; |