diff options
Diffstat (limited to 'modules/audio_remote_submix/audio_hw.cpp')
| -rw-r--r-- | modules/audio_remote_submix/audio_hw.cpp | 239 |
1 files changed, 25 insertions, 214 deletions
diff --git a/modules/audio_remote_submix/audio_hw.cpp b/modules/audio_remote_submix/audio_hw.cpp index b43a44dc..f96854b5 100644 --- a/modules/audio_remote_submix/audio_hw.cpp +++ b/modules/audio_remote_submix/audio_hw.cpp @@ -63,7 +63,7 @@ namespace android { #endif // SUBMIX_VERBOSE_LOGGING // NOTE: This value will be rounded up to the nearest power of 2 by MonoPipe(). -#define DEFAULT_PIPE_SIZE_IN_FRAMES (1024*4) +#define DEFAULT_PIPE_SIZE_IN_FRAMES (1024*4) // size at default sample rate // Value used to divide the MonoPipe() buffer into segments that are written to the source and // read from the sink. The maximum latency of the device is the size of the MonoPipe's buffer // the minimum latency is the MonoPipe buffer size divided by this value. @@ -83,10 +83,7 @@ namespace android { // multiple input streams from this device. If this option is enabled, each input stream returned // is *the same stream* which means that readers will race to read data from these streams. #define ENABLE_LEGACY_INPUT_OPEN 1 -// Whether channel conversion (16-bit signed PCM mono->stereo, stereo->mono) is enabled. -#define ENABLE_CHANNEL_CONVERSION 1 -// Whether resampling is enabled. -#define ENABLE_RESAMPLING 1 + #if LOG_STREAMS_TO_FILES // Folder to save stream log files to. #define LOG_STREAM_FOLDER "/data/misc/audioserver" @@ -130,11 +127,6 @@ struct submix_config { // channel bitfields are not equivalent. audio_channel_mask_t input_channel_mask; audio_channel_mask_t output_channel_mask; -#if ENABLE_RESAMPLING - // Input stream and output stream sample rates. - uint32_t input_sample_rate; - uint32_t output_sample_rate; -#endif // ENABLE_RESAMPLING size_t pipe_frame_size; // Number of bytes in each audio frame in the pipe. size_t buffer_size_frames; // Size of the audio pipe in frames. // Maximum number of frames buffered by the input and output streams. @@ -159,11 +151,6 @@ 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 - // stream. - int16_t resampler_buffer[DEFAULT_PIPE_SIZE_IN_FRAMES]; -#endif // ENABLE_RESAMPLING } route_config_t; struct submix_audio_device { @@ -221,6 +208,11 @@ static bool sample_rate_supported(const uint32_t sample_rate) return return_value; } +static size_t pipe_size_in_frames(const uint32_t sample_rate) +{ + return DEFAULT_PIPE_SIZE_IN_FRAMES * ((float) sample_rate / DEFAULT_SAMPLE_RATE_HZ); +} + // Determine whether the specified sample rate is supported, if it is return the specified sample // rate, otherwise return the default sample rate for the submix module. static uint32_t get_supported_sample_rate(uint32_t sample_rate) @@ -325,7 +317,6 @@ static struct submix_audio_device * audio_hw_device_get_submix_audio_device( static bool audio_config_compare(const audio_config * const input_config, const audio_config * const output_config) { -#if !ENABLE_CHANNEL_CONVERSION const uint32_t input_channels = audio_channel_count_from_in_mask(input_config->channel_mask); const uint32_t output_channels = audio_channel_count_from_out_mask(output_config->channel_mask); if (input_channels != output_channels) { @@ -333,13 +324,8 @@ static bool audio_config_compare(const audio_config * const input_config, input_channels, output_channels); return false; } -#endif // !ENABLE_CHANNEL_CONVERSION -#if ENABLE_RESAMPLING - if (input_config->sample_rate != output_config->sample_rate && - audio_channel_count_from_in_mask(input_config->channel_mask) != 1) { -#else + if (input_config->sample_rate != output_config->sample_rate) { -#endif // ENABLE_RESAMPLING ALOGE("audio_config_compare() sample rate mismatch %ul vs. %ul", input_config->sample_rate, output_config->sample_rate); return false; @@ -376,24 +362,11 @@ static void submix_audio_device_create_pipe_l(struct submix_audio_device * const in->route_handle = route_idx; rsxadev->routes[route_idx].input = in; rsxadev->routes[route_idx].config.input_channel_mask = config->channel_mask; -#if ENABLE_RESAMPLING - rsxadev->routes[route_idx].config.input_sample_rate = config->sample_rate; - // If the output isn't configured yet, set the output sample rate to the maximum supported - // sample rate such that the smallest possible input buffer is created, and put a default - // value for channel count - if (!rsxadev->routes[route_idx].output) { - rsxadev->routes[route_idx].config.output_sample_rate = 48000; - rsxadev->routes[route_idx].config.output_channel_mask = AUDIO_CHANNEL_OUT_STEREO; - } -#endif // ENABLE_RESAMPLING } if (out) { out->route_handle = route_idx; rsxadev->routes[route_idx].output = out; rsxadev->routes[route_idx].config.output_channel_mask = config->channel_mask; -#if ENABLE_RESAMPLING - rsxadev->routes[route_idx].config.output_sample_rate = config->sample_rate; -#endif // ENABLE_RESAMPLING } // Save the address strncpy(rsxadev->routes[route_idx].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN); @@ -403,18 +376,14 @@ static void submix_audio_device_create_pipe_l(struct submix_audio_device * const { struct submix_config * const device_config = &rsxadev->routes[route_idx].config; uint32_t channel_count; - if (out) + if (out) { channel_count = audio_channel_count_from_out_mask(config->channel_mask); - else + } else { channel_count = audio_channel_count_from_in_mask(config->channel_mask); -#if ENABLE_CHANNEL_CONVERSION - // If channel conversion is enabled, allocate enough space for the maximum number of - // possible channels stored in the pipe for the situation when the number of channels in - // the output stream don't match the number in the input stream. - const uint32_t pipe_channel_count = max(channel_count, 2); -#else + } + const uint32_t pipe_channel_count = channel_count; -#endif // ENABLE_CHANNEL_CONVERSION + const NBAIO_Format format = Format_from_SR_C(config->sample_rate, pipe_channel_count, config->format); const NBAIO_Format offers[1] = {format}; @@ -444,11 +413,7 @@ static void submix_audio_device_create_pipe_l(struct submix_audio_device * const buffer_period_count; if (in) device_config->pipe_frame_size = audio_stream_in_frame_size(&in->stream); if (out) device_config->pipe_frame_size = audio_stream_out_frame_size(&out->stream); -#if ENABLE_CHANNEL_CONVERSION - // Calculate the pipe frame size based upon the number of channels. - device_config->pipe_frame_size = (device_config->pipe_frame_size * pipe_channel_count) / - channel_count; -#endif // ENABLE_CHANNEL_CONVERSION + SUBMIX_ALOGV("submix_audio_device_create_pipe_l(): pipe frame size %zd, pipe size %zd, " "period size %zd", device_config->pipe_frame_size, device_config->buffer_size_frames, device_config->buffer_period_size_frames); @@ -473,10 +438,6 @@ static void submix_audio_device_release_pipe_l(struct submix_audio_device * cons rsxadev->routes[route_idx].rsxSource.clear(); } memset(rsxadev->routes[route_idx].address, 0, AUDIO_DEVICE_MAX_ADDRESS_LEN); -#if ENABLE_RESAMPLING - memset(rsxadev->routes[route_idx].resampler_buffer, 0, - sizeof(int16_t) * DEFAULT_PIPE_SIZE_IN_FRAMES); -#endif } // Remove references to the specified input and output streams. When the device no longer @@ -624,11 +585,7 @@ static uint32_t out_get_sample_rate(const struct audio_stream *stream) { const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( const_cast<struct audio_stream *>(stream)); -#if ENABLE_RESAMPLING - const uint32_t out_rate = out->dev->routes[out->route_handle].config.output_sample_rate; -#else const uint32_t out_rate = out->dev->routes[out->route_handle].config.common.sample_rate; -#endif // ENABLE_RESAMPLING SUBMIX_ALOGV("out_get_sample_rate() returns %u for addr %s", out_rate, out->dev->routes[out->route_handle].address); return out_rate; @@ -637,17 +594,6 @@ static uint32_t out_get_sample_rate(const struct audio_stream *stream) static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) { struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); -#if ENABLE_RESAMPLING - // The sample rate of the stream can't be changed once it's set since this would change the - // output buffer size and hence break playback to the shared pipe. - if (rate != out->dev->routes[out->route_handle].config.output_sample_rate) { - ALOGE("out_set_sample_rate() resampling enabled can't change sample rate from " - "%u to %u for addr %s", - out->dev->routes[out->route_handle].config.output_sample_rate, rate, - out->dev->routes[out->route_handle].address); - return -ENOSYS; - } -#endif // ENABLE_RESAMPLING if (!sample_rate_supported(rate)) { ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate); return -ENOSYS; @@ -994,11 +940,7 @@ static uint32_t in_get_sample_rate(const struct audio_stream *stream) { const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( const_cast<struct audio_stream*>(stream)); -#if ENABLE_RESAMPLING - const uint32_t rate = in->dev->routes[in->route_handle].config.input_sample_rate; -#else const uint32_t rate = in->dev->routes[in->route_handle].config.common.sample_rate; -#endif // ENABLE_RESAMPLING SUBMIX_ALOGV("in_get_sample_rate() returns %u", rate); return rate; } @@ -1006,15 +948,6 @@ static uint32_t in_get_sample_rate(const struct audio_stream *stream) static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) { const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); -#if ENABLE_RESAMPLING - // The sample rate of the stream can't be changed once it's set since this would change the - // input buffer size and hence break recording from the shared pipe. - if (rate != in->dev->routes[in->route_handle].config.input_sample_rate) { - ALOGE("in_set_sample_rate() resampling enabled can't change sample rate from " - "%u to %u", in->dev->routes[in->route_handle].config.input_sample_rate, rate); - return -ENOSYS; - } -#endif // ENABLE_RESAMPLING if (!sample_rate_supported(rate)) { ALOGE("in_set_sample_rate(rate=%u) rate unsupported", rate); return -ENOSYS; @@ -1033,13 +966,6 @@ static size_t in_get_buffer_size(const struct audio_stream *stream) audio_stream_in_frame_size((const struct audio_stream_in *)stream); size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( stream, config, config->buffer_period_size_frames, stream_frame_size); -#if ENABLE_RESAMPLING - // Scale the size of the buffer based upon the maximum number of frames that could be returned - // given the ratio of output to input sample rate. - buffer_size_frames = (size_t)(((float)buffer_size_frames * - (float)config->input_sample_rate) / - (float)config->output_sample_rate); -#endif // ENABLE_RESAMPLING const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size; SUBMIX_ALOGV("in_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes, buffer_size_frames); @@ -1168,65 +1094,10 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, // read the data from the pipe (it's non blocking) int attempts = 0; char* buff = (char*)buffer; -#if ENABLE_CHANNEL_CONVERSION - // Determine whether channel conversion is required. - const uint32_t input_channels = audio_channel_count_from_in_mask( - rsxadev->routes[in->route_handle].config.input_channel_mask); - const uint32_t output_channels = audio_channel_count_from_out_mask( - rsxadev->routes[in->route_handle].config.output_channel_mask); - if (input_channels != output_channels) { - SUBMIX_ALOGV("in_read(): %d output channels will be converted to %d " - "input channels", output_channels, input_channels); - // Only support 16-bit PCM channel conversion from mono to stereo or stereo to mono. - ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format == - AUDIO_FORMAT_PCM_16_BIT); - ALOG_ASSERT((input_channels == 1 && output_channels == 2) || - (input_channels == 2 && output_channels == 1)); - } -#endif // ENABLE_CHANNEL_CONVERSION - -#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) { - resampler_ratio = (float)output_sample_rate / (float)input_sample_rate; - // Only support 16-bit PCM mono resampling. - // NOTE: Resampling is performed after the channel conversion step. - ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format == - AUDIO_FORMAT_PCM_16_BIT); - ALOG_ASSERT(audio_channel_count_from_in_mask( - rsxadev->routes[in->route_handle].config.input_channel_mask) == 1); - } -#endif // ENABLE_RESAMPLING while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) { ssize_t frames_read = -1977; size_t read_frames = remaining_frames; -#if ENABLE_RESAMPLING - char* const saved_buff = buff; - 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 into the resampler buffer. - buff = (char*)rsxadev->routes[in->route_handle].resampler_buffer; - } -#endif // ENABLE_RESAMPLING -#if ENABLE_CHANNEL_CONVERSION - if (output_channels == 1 && input_channels == 2) { - // Need to read half the requested frames since the converted output - // data will take twice the space (mono->stereo). - read_frames /= 2; - } -#endif // ENABLE_CHANNEL_CONVERSION SUBMIX_ALOGV("in_read(): frames available to read %zd", source->availableToRead()); @@ -1234,56 +1105,6 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, SUBMIX_ALOGV("in_read(): frames read %zd", frames_read); -#if ENABLE_CHANNEL_CONVERSION - // Perform in-place channel conversion. - // NOTE: In the following "input stream" refers to the data returned by this function - // and "output stream" refers to the data read from the pipe. - if (input_channels != output_channels && frames_read > 0) { - int16_t *data = (int16_t*)buff; - if (output_channels == 2 && input_channels == 1) { - // Offset into the output stream data in samples. - ssize_t output_stream_offset = 0; - 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; - } - } else if (output_channels == 1 && input_channels == 2) { - // Offset into the input stream data in samples. - ssize_t input_stream_offset = (frames_read - 1) * 2; - for (ssize_t output_stream_frame = frames_read - 1; output_stream_frame >= 0; - output_stream_frame--, input_stream_offset -= 2) { - const short sample = data[output_stream_frame]; - data[input_stream_offset] = sample; - data[input_stream_offset + 1] = sample; - } - } - } -#endif // ENABLE_CHANNEL_CONVERSION - -#if ENABLE_RESAMPLING - if (resampler_ratio != 1.0f) { - SUBMIX_ALOGV("in_read(): resampling %zd frames", frames_read); - const int16_t * const data = (int16_t*)buff; - int16_t * const resampled_buffer = (int16_t*)saved_buff; - // Resample with *no* filtering - if the data from the ouptut stream was really - // 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; - 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]; - } - ALOG_ASSERT(input_stream_frame <= (ssize_t)resampler_buffer_size_frames); - SUBMIX_ALOGV("in_read(): resampler produced %zd frames", input_stream_frame); - frames_read = input_stream_frame; - buff = saved_buff; - } -#endif // ENABLE_RESAMPLING - if (frames_read > 0) { #if LOG_STREAMS_TO_FILES if (in->log_fd >= 0) write(in->log_fd, buff, frames_read * frame_size); @@ -1411,7 +1232,6 @@ static int adev_open_output_stream(struct audio_hw_device *dev, struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); ALOGD("adev_open_output_stream(address=%s)", address); struct submix_stream_out *out; - bool force_pipe_creation = false; (void)handle; (void)devices; (void)flags; @@ -1464,25 +1284,20 @@ static int adev_open_output_stream(struct audio_hw_device *dev, out->stream.get_next_write_timestamp = out_get_next_write_timestamp; out->stream.get_presentation_position = out_get_presentation_position; -#if ENABLE_RESAMPLING - // Recreate the pipe with the correct sample rate so that MonoPipe.write() rate limits - // writes correctly. - force_pipe_creation = rsxadev->routes[route_idx].config.common.sample_rate - != config->sample_rate; -#endif // ENABLE_RESAMPLING - // If the sink has been shutdown or pipe recreation is forced (see above), delete the pipe so // that it's recreated. if ((rsxadev->routes[route_idx].rsxSink != NULL - && rsxadev->routes[route_idx].rsxSink->isShutdown()) || force_pipe_creation) { + && rsxadev->routes[route_idx].rsxSink->isShutdown())) { submix_audio_device_release_pipe_l(rsxadev, route_idx); } // Store a pointer to the device from the output stream. out->dev = rsxadev; // Initialize the pipe. - ALOGV("adev_open_output_stream(): about to create pipe at index %d", route_idx); - submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, + const size_t pipeSizeInFrames = pipe_size_in_frames(config->sample_rate); + ALOGI("adev_open_output_stream(): about to create pipe at index %d, rate %u, pipe size %zu", + route_idx, config->sample_rate, pipeSizeInFrames); + submix_audio_device_create_pipe_l(rsxadev, config, pipeSizeInFrames, DEFAULT_PIPE_PERIOD_COUNT, NULL, out, address, route_idx); #if LOG_STREAMS_TO_FILES out->log_fd = open(LOG_STREAM_OUT_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, @@ -1611,7 +1426,8 @@ static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, const size_t frame_size_in_bytes = audio_channel_count_from_in_mask(config->channel_mask) * audio_bytes_per_sample(config->format); if (max_buffer_period_size_frames == 0) { - max_buffer_period_size_frames = DEFAULT_PIPE_SIZE_IN_FRAMES; + max_buffer_period_size_frames = + pipe_size_in_frames(get_supported_sample_rate(config->sample_rate));; } const size_t buffer_size = max_buffer_period_size_frames * frame_size_in_bytes; SUBMIX_ALOGV("adev_get_input_buffer_size() returns %zu bytes, %zu frames", @@ -1724,8 +1540,10 @@ static int adev_open_input_stream(struct audio_hw_device *dev, in->read_error_count = 0; // Initialize the pipe. - ALOGV("adev_open_input_stream(): about to create pipe"); - submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, + const size_t pipeSizeInFrames = pipe_size_in_frames(config->sample_rate); + ALOGI("adev_open_input_stream(): about to create pipe at index %d, rate %u, pipe size %zu", + route_idx, config->sample_rate, pipeSizeInFrames); + submix_audio_device_create_pipe_l(rsxadev, config, pipeSizeInFrames, DEFAULT_PIPE_PERIOD_COUNT, in, NULL, address, route_idx); sp <MonoPipe> sink = rsxadev->routes[route_idx].rsxSink; @@ -1779,16 +1597,9 @@ static int adev_dump(const audio_hw_device_t *device, int fd) int n = snprintf(msg, sizeof(msg), "\nReroute submix audio module:\n"); write(fd, &msg, n); for (int i=0 ; i < MAX_ROUTES ; i++) { -#if ENABLE_RESAMPLING - n = snprintf(msg, sizeof(msg), " route[%d] rate in=%d out=%d, addr=[%s]\n", i, - rsxadev->routes[i].config.input_sample_rate, - rsxadev->routes[i].config.output_sample_rate, - rsxadev->routes[i].address); -#else n = snprintf(msg, sizeof(msg), " route[%d], rate=%d addr=[%s]\n", i, rsxadev->routes[i].config.common.sample_rate, rsxadev->routes[i].address); -#endif write(fd, &msg, n); } return 0; |