diff options
Diffstat (limited to 'modules')
| -rw-r--r-- | modules/audio_remote_submix/audio_hw.cpp | 436 | ||||
| -rw-r--r-- | modules/sensors/multihal.cpp | 58 | ||||
| -rw-r--r-- | modules/usbaudio/alsa_device_profile.c | 42 | ||||
| -rw-r--r-- | modules/usbaudio/audio_hw.c | 55 |
4 files changed, 391 insertions, 200 deletions
diff --git a/modules/audio_remote_submix/audio_hw.cpp b/modules/audio_remote_submix/audio_hw.cpp index 8fed8e4..b9dcf7a 100644 --- a/modules/audio_remote_submix/audio_hw.cpp +++ b/modules/audio_remote_submix/audio_hw.cpp @@ -140,11 +140,10 @@ struct submix_config { size_t buffer_period_size_frames; }; -struct submix_audio_device { - struct audio_hw_device device; - bool input_standby; - bool output_standby; - submix_config config; +#define MAX_ROUTES 10 +typedef struct route_config { + struct submix_config config; + char address[AUDIO_DEVICE_MAX_ADDRESS_LEN]; // Pipe variables: they handle the ring buffer that "pipes" audio: // - from the submix virtual audio output == what needs to be played // remotely, seen as an output for AudioFlinger @@ -155,17 +154,20 @@ struct submix_audio_device { // TV with Wifi Display capabilities), or to a wireless audio player. sp<MonoPipe> rsxSink; sp<MonoPipeReader> rsxSource; + // Pointers to the current input and output stream instances. rsxSink and rsxSource are + // 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; - // Pointers to the current input and output stream instances. rsxSink and rsxSource are - // destroyed if both and input and output streams are destroyed. - struct submix_stream_out *output; - struct submix_stream_in *input; - +struct submix_audio_device { + struct audio_hw_device device; + route_config_t routes[MAX_ROUTES]; // Device lock, also used to protect access to submix_audio_device from the input and output // streams. pthread_mutex_t lock; @@ -174,6 +176,8 @@ struct submix_audio_device { struct submix_stream_out { struct audio_stream_out stream; struct submix_audio_device *dev; + int route_handle; + bool output_standby; #if LOG_STREAMS_TO_FILES int log_fd; #endif // LOG_STREAMS_TO_FILES @@ -182,7 +186,9 @@ struct submix_stream_out { struct submix_stream_in { struct audio_stream_in stream; struct submix_audio_device *dev; - bool output_standby; // output standby state as seen from record thread + int route_handle; + bool input_standby; + bool output_standby_rec_thr; // output standby state as seen from record thread // wall clock when recording starts struct timespec record_start_time; @@ -346,40 +352,53 @@ static bool audio_config_compare(const audio_config * const input_config, // If one doesn't exist, create a pipe for the submix audio device rsxadev of size // buffer_size_frames and optionally associate "in" or "out" with the submix audio device. -static void submix_audio_device_create_pipe(struct submix_audio_device * const rsxadev, +// Must be called with lock held on the submix_audio_device +static void submix_audio_device_create_pipe_l(struct submix_audio_device * const rsxadev, const struct audio_config * const config, const size_t buffer_size_frames, const uint32_t buffer_period_count, struct submix_stream_in * const in, - struct submix_stream_out * const out) + struct submix_stream_out * const out, + const char *address, + int route_idx) { ALOG_ASSERT(in || out); - ALOGD("submix_audio_device_create_pipe()"); - pthread_mutex_lock(&rsxadev->lock); + ALOG_ASSERT(route_idx > -1); + ALOG_ASSERT(route_idx < MAX_ROUTES); + ALOGD("submix_audio_device_create_pipe_l(addr=%s, idx=%d)", address, route_idx); + // Save a reference to the specified input or output stream and the associated channel // mask. if (in) { - rsxadev->input = in; - rsxadev->config.input_channel_mask = config->channel_mask; + 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->config.input_sample_rate = config->sample_rate; + 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. - if (!rsxadev->output) { - rsxadev->config.output_sample_rate = 48000; + // 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) { - rsxadev->output = out; - rsxadev->config.output_channel_mask = config->channel_mask; + 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->config.output_sample_rate = config->sample_rate; + 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); + ALOGD(" now using address %s for route %d", rsxadev->routes[route_idx].address, route_idx); // If a pipe isn't associated with the device, create one. - if (rsxadev->rsxSink == NULL || rsxadev->rsxSource == NULL) { - struct submix_config * const device_config = &rsxadev->config; + if (rsxadev->routes[route_idx].rsxSink == NULL || rsxadev->routes[route_idx].rsxSource == NULL) + { + struct submix_config * const device_config = &rsxadev->routes[route_idx].config; uint32_t channel_count; if (out) channel_count = audio_channel_count_from_out_mask(config->channel_mask); @@ -407,13 +426,13 @@ static void submix_audio_device_create_pipe(struct submix_audio_device * const r numCounterOffers = 0; index = source->negotiate(offers, 1, NULL, numCounterOffers); ALOG_ASSERT(index == 0); - ALOGV("submix_audio_device_create_pipe(): created pipe"); + ALOGV("submix_audio_device_create_pipe_l(): created pipe"); // Save references to the source and sink. - ALOG_ASSERT(rsxadev->rsxSink == NULL); - ALOG_ASSERT(rsxadev->rsxSource == NULL); - rsxadev->rsxSink = sink; - rsxadev->rsxSource = source; + ALOG_ASSERT(rsxadev->routes[route_idx].rsxSink == NULL); + ALOG_ASSERT(rsxadev->routes[route_idx].rsxSource == NULL); + rsxadev->routes[route_idx].rsxSink = sink; + rsxadev->routes[route_idx].rsxSource = source; // Store the sanitized audio format in the device so that it's possible to determine // the format of the pipe source when opening the input device. memcpy(&device_config->common, config, sizeof(device_config->common)); @@ -427,51 +446,71 @@ static void submix_audio_device_create_pipe(struct submix_audio_device * const r 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(): pipe frame size %zd, pipe size %zd, " + 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); } - pthread_mutex_unlock(&rsxadev->lock); } // Release references to the sink and source. Input and output threads may maintain references // to these objects via StrongPointer (sp<MonoPipe> and sp<MonoPipeReader>) which they can use // before they shutdown. -static void submix_audio_device_release_pipe(struct submix_audio_device * const rsxadev) -{ - ALOGD("submix_audio_device_release_pipe()"); - rsxadev->rsxSink.clear(); - rsxadev->rsxSource.clear(); +// Must be called with lock held on the submix_audio_device +static void submix_audio_device_release_pipe_l(struct submix_audio_device * const rsxadev, + int route_idx) +{ + ALOG_ASSERT(route_idx > -1); + ALOG_ASSERT(route_idx < MAX_ROUTES); + ALOGD("submix_audio_device_release_pipe_l(idx=%d) addr=%s", route_idx, + rsxadev->routes[route_idx].address); + if (rsxadev->routes[route_idx].rsxSink != 0) { + rsxadev->routes[route_idx].rsxSink.clear(); + rsxadev->routes[route_idx].rsxSink = 0; + } + if (rsxadev->routes[route_idx].rsxSource != 0) { + rsxadev->routes[route_idx].rsxSource.clear(); + 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, + sizeof(int16_t) * DEFAULT_PIPE_SIZE_IN_FRAMES); +#endif } // Remove references to the specified input and output streams. When the device no longer // references input and output streams destroy the associated pipe. -static void submix_audio_device_destroy_pipe(struct submix_audio_device * const rsxadev, +// Must be called with lock held on the submix_audio_device +static void submix_audio_device_destroy_pipe_l(struct submix_audio_device * const rsxadev, const struct submix_stream_in * const in, const struct submix_stream_out * const out) { MonoPipe* sink; - pthread_mutex_lock(&rsxadev->lock); - ALOGV("submix_audio_device_destroy_pipe()"); - ALOG_ASSERT(in == NULL || rsxadev->input == in); - ALOG_ASSERT(out == NULL || rsxadev->output == out); + ALOGV("submix_audio_device_destroy_pipe_l()"); + int route_idx = -1; if (in != NULL) { #if ENABLE_LEGACY_INPUT_OPEN const_cast<struct submix_stream_in*>(in)->ref_count--; + route_idx = in->route_handle; + ALOG_ASSERT(rsxadev->routes[route_idx].input == in); if (in->ref_count == 0) { - rsxadev->input = NULL; + rsxadev->routes[route_idx].input = NULL; } - ALOGV("submix_audio_device_destroy_pipe(): input ref_count %d", in->ref_count); + ALOGV("submix_audio_device_destroy_pipe_l(): input ref_count %d", in->ref_count); #else rsxadev->input = NULL; #endif // ENABLE_LEGACY_INPUT_OPEN } - if (out != NULL) rsxadev->output = NULL; - if (rsxadev->input == NULL && rsxadev->output == NULL) { - submix_audio_device_release_pipe(rsxadev); - ALOGD("submix_audio_device_destroy_pipe(): pipe destroyed"); + if (out != NULL) { + route_idx = out->route_handle; + ALOG_ASSERT(rsxadev->routes[route_idx].output == out); + rsxadev->routes[route_idx].output = NULL; + } + if (route_idx != -1 && + rsxadev->routes[route_idx].input == NULL && rsxadev->routes[route_idx].output == NULL) { + submix_audio_device_release_pipe_l(rsxadev, route_idx); + ALOGD("submix_audio_device_destroy_pipe_l(): pipe destroyed"); } - pthread_mutex_unlock(&rsxadev->lock); } // Sanitize the user specified audio config for a submix input / output stream. @@ -484,8 +523,9 @@ static void submix_sanitize_config(struct audio_config * const config, const boo } // Verify a submix input or output stream can be opened. -static bool submix_open_validate(const struct submix_audio_device * const rsxadev, - pthread_mutex_t * const lock, +// Must be called with lock held on the submix_audio_device +static bool submix_open_validate_l(const struct submix_audio_device * const rsxadev, + int route_idx, const struct audio_config * const config, const bool opening_input) { @@ -494,20 +534,18 @@ static bool submix_open_validate(const struct submix_audio_device * const rsxade audio_config pipe_config; // Query the device for the current audio config and whether input and output streams are open. - pthread_mutex_lock(lock); - output_open = rsxadev->output != NULL; - input_open = rsxadev->input != NULL; - memcpy(&pipe_config, &rsxadev->config.common, sizeof(pipe_config)); - pthread_mutex_unlock(lock); + output_open = rsxadev->routes[route_idx].output != NULL; + input_open = rsxadev->routes[route_idx].input != NULL; + memcpy(&pipe_config, &rsxadev->routes[route_idx].config.common, sizeof(pipe_config)); // If the stream is already open, don't open it again. if (opening_input ? !ENABLE_LEGACY_INPUT_OPEN && input_open : output_open) { - ALOGE("submix_open_validate(): %s stream already open.", opening_input ? "Input" : + ALOGE("submix_open_validate_l(): %s stream already open.", opening_input ? "Input" : "Output"); return false; } - SUBMIX_ALOGV("submix_open_validate(): sample rate=%d format=%x " + SUBMIX_ALOGV("submix_open_validate_l(): sample rate=%d format=%x " "%s_channel_mask=%x", config->sample_rate, config->format, opening_input ? "in" : "out", config->channel_mask); @@ -518,16 +556,46 @@ static bool submix_open_validate(const struct submix_audio_device * const rsxade const audio_config * const output_config = opening_input ? &pipe_config : config; // Get the channel mask of the open device. pipe_config.channel_mask = - opening_input ? rsxadev->config.output_channel_mask : - rsxadev->config.input_channel_mask; + opening_input ? rsxadev->routes[route_idx].config.output_channel_mask : + rsxadev->routes[route_idx].config.input_channel_mask; if (!audio_config_compare(input_config, output_config)) { - ALOGE("submix_open_validate(): Unsupported format."); + ALOGE("submix_open_validate_l(): Unsupported format."); return false; } } return true; } +// Must be called with lock held on the submix_audio_device +static status_t submix_get_route_idx_for_address_l(const struct submix_audio_device * const rsxadev, + const char* address, /*in*/ + int *idx /*out*/) +{ + // Do we already have a route for this address + int route_idx = -1; + int route_empty_idx = -1; // index of an empty route slot that can be used if needed + for (int i=0 ; i < MAX_ROUTES ; i++) { + if (strcmp(rsxadev->routes[i].address, "") == 0) { + route_empty_idx = i; + } + if (strncmp(rsxadev->routes[i].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) { + route_idx = i; + break; + } + } + + if ((route_idx == -1) && (route_empty_idx == -1)) { + ALOGE("Cannot create new route for address %s, max number of routes reached", address); + return -ENOMEM; + } + if (route_idx == -1) { + route_idx = route_empty_idx; + } + *idx = route_idx; + return OK; +} + + // Calculate the maximum size of the pipe buffer in frames for the specified stream. static size_t calculate_stream_pipe_size_in_frames(const struct audio_stream *stream, const struct submix_config *config, @@ -546,11 +614,12 @@ 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->config.output_sample_rate; + const uint32_t out_rate = out->dev->routes[out->route_handle].config.output_sample_rate; #else - const uint32_t out_rate = out->dev->config.common.sample_rate; + 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", out_rate); + SUBMIX_ALOGV("out_get_sample_rate() returns %u for addr %s", + out_rate, out->dev->routes[out->route_handle].address); return out_rate; } @@ -560,9 +629,11 @@ static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) #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->config.output_sample_rate) { + 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", out->dev->config.output_sample_rate, rate); + "%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 @@ -571,7 +642,7 @@ static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) return -ENOSYS; } SUBMIX_ALOGV("out_set_sample_rate(rate=%u)", rate); - out->dev->config.common.sample_rate = rate; + out->dev->routes[out->route_handle].config.common.sample_rate = rate; return 0; } @@ -579,7 +650,7 @@ static size_t out_get_buffer_size(const struct audio_stream *stream) { const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( const_cast<struct audio_stream *>(stream)); - const struct submix_config * const config = &out->dev->config; + const struct submix_config * const config = &out->dev->routes[out->route_handle].config; const size_t stream_frame_size = audio_stream_out_frame_size((const struct audio_stream_out *)stream); const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( @@ -594,7 +665,7 @@ static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) { const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( const_cast<struct audio_stream *>(stream)); - uint32_t channel_mask = out->dev->config.output_channel_mask; + uint32_t channel_mask = out->dev->routes[out->route_handle].config.output_channel_mask; SUBMIX_ALOGV("out_get_channels() returns %08x", channel_mask); return channel_mask; } @@ -603,7 +674,7 @@ static audio_format_t out_get_format(const struct audio_stream *stream) { const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( const_cast<struct audio_stream *>(stream)); - const audio_format_t format = out->dev->config.common.format; + const audio_format_t format = out->dev->routes[out->route_handle].config.common.format; SUBMIX_ALOGV("out_get_format() returns %x", format); return format; } @@ -611,7 +682,7 @@ static audio_format_t out_get_format(const struct audio_stream *stream) static int out_set_format(struct audio_stream *stream, audio_format_t format) { const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); - if (format != out->dev->config.common.format) { + if (format != out->dev->routes[out->route_handle].config.common.format) { ALOGE("out_set_format(format=%x) format unsupported", format); return -ENOSYS; } @@ -621,12 +692,13 @@ static int out_set_format(struct audio_stream *stream, audio_format_t format) static int out_standby(struct audio_stream *stream) { - struct submix_audio_device * const rsxadev = audio_stream_get_submix_stream_out(stream)->dev; ALOGI("out_standby()"); + struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); + struct submix_audio_device * const rsxadev = out->dev; pthread_mutex_lock(&rsxadev->lock); - rsxadev->output_standby = true; + out->output_standby = true; pthread_mutex_unlock(&rsxadev->lock); @@ -653,7 +725,9 @@ static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) audio_stream_get_submix_stream_out(stream)->dev; pthread_mutex_lock(&rsxadev->lock); { // using the sink - sp<MonoPipe> sink = rsxadev->rsxSink; + sp<MonoPipe> sink = + rsxadev->routes[audio_stream_get_submix_stream_out(stream)->route_handle] + .rsxSink; if (sink == NULL) { pthread_mutex_unlock(&rsxadev->lock); return 0; @@ -678,7 +752,7 @@ static uint32_t out_get_latency(const struct audio_stream_out *stream) { const struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out( const_cast<struct audio_stream_out *>(stream)); - const struct submix_config * const config = &out->dev->config; + const struct submix_config * const config = &out->dev->routes[out->route_handle].config; const size_t stream_frame_size = audio_stream_out_frame_size(stream); const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( @@ -711,9 +785,9 @@ static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, pthread_mutex_lock(&rsxadev->lock); - rsxadev->output_standby = false; + out->output_standby = false; - sp<MonoPipe> sink = rsxadev->rsxSink; + sp<MonoPipe> sink = rsxadev->routes[out->route_handle].rsxSink; if (sink != NULL) { if (sink->isShutdown()) { sink.clear(); @@ -735,8 +809,8 @@ static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, // from the pipe to make space to write the most recent data. { const size_t availableToWrite = sink->availableToWrite(); - sp<MonoPipeReader> source = rsxadev->rsxSource; - if (rsxadev->input == NULL && availableToWrite < frames) { + sp<MonoPipeReader> source = rsxadev->routes[out->route_handle].rsxSource; + if (rsxadev->routes[out->route_handle].input == NULL && availableToWrite < frames) { static uint8_t flush_buffer[64]; const size_t flushBufferSizeFrames = sizeof(flush_buffer) / frame_size; size_t frames_to_flush_from_source = frames - availableToWrite; @@ -745,6 +819,7 @@ static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, while (frames_to_flush_from_source) { const size_t flush_size = min(frames_to_flush_from_source, flushBufferSizeFrames); frames_to_flush_from_source -= flush_size; + // read does not block source->read(flush_buffer, flush_size, AudioBufferProvider::kInvalidPTS); } } @@ -824,9 +899,9 @@ 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->config.input_sample_rate; + const uint32_t rate = in->dev->routes[in->route_handle].config.input_sample_rate; #else - const uint32_t rate = in->dev->config.common.sample_rate; + 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; @@ -838,9 +913,9 @@ static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) #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->config.input_sample_rate) { + 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->config.input_sample_rate, rate); + "%u to %u", in->dev->routes[in->route_handle].config.input_sample_rate, rate); return -ENOSYS; } #endif // ENABLE_RESAMPLING @@ -848,7 +923,7 @@ static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) ALOGE("in_set_sample_rate(rate=%u) rate unsupported", rate); return -ENOSYS; } - in->dev->config.common.sample_rate = rate; + in->dev->routes[in->route_handle].config.common.sample_rate = rate; SUBMIX_ALOGV("in_set_sample_rate() set %u", rate); return 0; } @@ -857,7 +932,7 @@ static size_t in_get_buffer_size(const struct audio_stream *stream) { const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( const_cast<struct audio_stream*>(stream)); - const struct submix_config * const config = &in->dev->config; + const struct submix_config * const config = &in->dev->routes[in->route_handle].config; const size_t stream_frame_size = audio_stream_in_frame_size((const struct audio_stream_in *)stream); size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( @@ -879,7 +954,8 @@ static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) { const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( const_cast<struct audio_stream*>(stream)); - const audio_channel_mask_t channel_mask = in->dev->config.input_channel_mask; + const audio_channel_mask_t channel_mask = + in->dev->routes[in->route_handle].config.input_channel_mask; SUBMIX_ALOGV("in_get_channels() returns %x", channel_mask); return channel_mask; } @@ -888,7 +964,7 @@ static audio_format_t in_get_format(const struct audio_stream *stream) { const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( const_cast<struct audio_stream*>(stream)); - const audio_format_t format = in->dev->config.common.format; + const audio_format_t format = in->dev->routes[in->route_handle].config.common.format; SUBMIX_ALOGV("in_get_format() returns %x", format); return format; } @@ -896,7 +972,7 @@ static audio_format_t in_get_format(const struct audio_stream *stream) static int in_set_format(struct audio_stream *stream, audio_format_t format) { const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); - if (format != in->dev->config.common.format) { + if (format != in->dev->routes[in->route_handle].config.common.format) { ALOGE("in_set_format(format=%x) format unsupported", format); return -ENOSYS; } @@ -906,12 +982,13 @@ static int in_set_format(struct audio_stream *stream, audio_format_t format) static int in_standby(struct audio_stream *stream) { - struct submix_audio_device * const rsxadev = audio_stream_get_submix_stream_in(stream)->dev; ALOGI("in_standby()"); + struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); + struct submix_audio_device * const rsxadev = in->dev; pthread_mutex_lock(&rsxadev->lock); - rsxadev->input_standby = true; + in->input_standby = true; pthread_mutex_unlock(&rsxadev->lock); @@ -959,11 +1036,13 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, SUBMIX_ALOGV("in_read bytes=%zu", bytes); pthread_mutex_lock(&rsxadev->lock); - const bool output_standby_transition = (in->output_standby != in->dev->output_standby); - in->output_standby = rsxadev->output_standby; + const bool output_standby = rsxadev->routes[in->route_handle].output == NULL + ? true : rsxadev->routes[in->route_handle].output->output_standby; + const bool output_standby_transition = (in->output_standby_rec_thr != output_standby); + in->output_standby_rec_thr = output_standby; - if (rsxadev->input_standby || output_standby_transition) { - rsxadev->input_standby = false; + if (in->input_standby || output_standby_transition) { + in->input_standby = false; // keep track of when we exit input standby (== first read == start "real recording") // or when we start recording silence, and reset projected time int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time); @@ -977,7 +1056,7 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, { // about to read from audio source - sp<MonoPipeReader> source = rsxadev->rsxSource; + sp<MonoPipeReader> source = rsxadev->routes[in->route_handle].rsxSource; if (source == NULL) { in->read_error_count++;// ok if it rolls over ALOGE_IF(in->read_error_count < MAX_READ_ERROR_LOGS, @@ -996,14 +1075,15 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, #if ENABLE_CHANNEL_CONVERSION // Determine whether channel conversion is required. const uint32_t input_channels = audio_channel_count_from_in_mask( - rsxadev->config.input_channel_mask); + rsxadev->routes[in->route_handle].config.input_channel_mask); const uint32_t output_channels = audio_channel_count_from_out_mask( - rsxadev->config.output_channel_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->config.common.format == AUDIO_FORMAT_PCM_16_BIT); + 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)); } @@ -1011,17 +1091,21 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, #if ENABLE_RESAMPLING const uint32_t input_sample_rate = in_get_sample_rate(&stream->common); - const uint32_t output_sample_rate = rsxadev->config.output_sample_rate; + const uint32_t output_sample_rate = + rsxadev->routes[in->route_handle].config.output_sample_rate; const size_t resampler_buffer_size_frames = - sizeof(rsxadev->resampler_buffer) / sizeof(rsxadev->resampler_buffer[0]); + 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->config.common.format == AUDIO_FORMAT_PCM_16_BIT); - ALOG_ASSERT(audio_channel_count_from_in_mask(rsxadev->config.input_channel_mask) == 1); + 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 @@ -1037,7 +1121,7 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, (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->resampler_buffer; + buff = (char*)rsxadev->routes[in->route_handle].resampler_buffer; } #endif // ENABLE_RESAMPLING #if ENABLE_CHANNEL_CONVERSION @@ -1195,10 +1279,10 @@ static int adev_open_output_stream(struct audio_hw_device *dev, audio_output_flags_t flags, struct audio_config *config, struct audio_stream_out **stream_out, - const char *address __unused) + const char *address) { struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); - ALOGD("adev_open_output_stream()"); + ALOGD("adev_open_output_stream(address=%s)", address); struct submix_stream_out *out; bool force_pipe_creation = false; (void)handle; @@ -1209,13 +1293,29 @@ static int adev_open_output_stream(struct audio_hw_device *dev, // Make sure it's possible to open the device given the current audio config. submix_sanitize_config(config, false); - if (!submix_open_validate(rsxadev, &rsxadev->lock, config, false)) { - ALOGE("adev_open_output_stream(): Unable to open output stream."); + + int route_idx = -1; + + pthread_mutex_lock(&rsxadev->lock); + + status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx); + if (res != OK) { + ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address); + pthread_mutex_unlock(&rsxadev->lock); + return res; + } + + if (!submix_open_validate_l(rsxadev, route_idx, config, false)) { + ALOGE("adev_open_output_stream(): Unable to open output stream for address %s", address); + pthread_mutex_unlock(&rsxadev->lock); return -EINVAL; } out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out)); - if (!out) return -ENOMEM; + if (!out) { + pthread_mutex_unlock(&rsxadev->lock); + return -ENOMEM; + } // Initialize the function pointer tables (v-tables). out->stream.common.get_sample_rate = out_get_sample_rate; @@ -1239,23 +1339,23 @@ static int adev_open_output_stream(struct audio_hw_device *dev, #if ENABLE_RESAMPLING // Recreate the pipe with the correct sample rate so that MonoPipe.write() rate limits // writes correctly. - force_pipe_creation = rsxadev->config.common.sample_rate != config->sample_rate; + 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. - pthread_mutex_lock(&rsxadev->lock); - if ((rsxadev->rsxSink != NULL && rsxadev->rsxSink->isShutdown()) || force_pipe_creation) { - submix_audio_device_release_pipe(rsxadev); + if ((rsxadev->routes[route_idx].rsxSink != NULL + && rsxadev->routes[route_idx].rsxSink->isShutdown()) || force_pipe_creation) { + submix_audio_device_release_pipe_l(rsxadev, route_idx); } - pthread_mutex_unlock(&rsxadev->lock); // 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"); - submix_audio_device_create_pipe(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, - DEFAULT_PIPE_PERIOD_COUNT, NULL, out); + 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, + 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, LOG_STREAM_FILE_PERMISSIONS); @@ -1266,18 +1366,25 @@ static int adev_open_output_stream(struct audio_hw_device *dev, // Return the output stream. *stream_out = &out->stream; + pthread_mutex_unlock(&rsxadev->lock); return 0; } static void adev_close_output_stream(struct audio_hw_device *dev, struct audio_stream_out *stream) { + struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device( + const_cast<struct audio_hw_device*>(dev)); struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); - ALOGD("adev_close_output_stream()"); - submix_audio_device_destroy_pipe(audio_hw_device_get_submix_audio_device(dev), NULL, out); + + pthread_mutex_lock(&rsxadev->lock); + ALOGD("adev_close_output_stream() addr = %s", rsxadev->routes[out->route_handle].address); + submix_audio_device_destroy_pipe_l(audio_hw_device_get_submix_audio_device(dev), NULL, out); #if LOG_STREAMS_TO_FILES if (out->log_fd >= 0) close(out->log_fd); #endif // LOG_STREAMS_TO_FILES + + pthread_mutex_unlock(&rsxadev->lock); free(out); } @@ -1363,12 +1470,19 @@ static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, const struct audio_config *config) { if (audio_is_linear_pcm(config->format)) { - const size_t buffer_period_size_frames = - audio_hw_device_get_submix_audio_device(const_cast<struct audio_hw_device*>(dev))-> - config.buffer_period_size_frames; + size_t max_buffer_period_size_frames = 0; + struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device( + const_cast<struct audio_hw_device*>(dev)); + // look for the largest buffer period size + for (int i = 0 ; i < MAX_ROUTES ; i++) { + if (rsxadev->routes[i].config.buffer_period_size_frames > max_buffer_period_size_frames) + { + max_buffer_period_size_frames = rsxadev->routes[i].config.buffer_period_size_frames; + } + } const size_t frame_size_in_bytes = audio_channel_count_from_in_mask(config->channel_mask) * audio_bytes_per_sample(config->format); - const size_t buffer_size = buffer_period_size_frames * frame_size_in_bytes; + 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", buffer_size, buffer_period_size_frames); return buffer_size; @@ -1382,37 +1496,49 @@ static int adev_open_input_stream(struct audio_hw_device *dev, struct audio_config *config, struct audio_stream_in **stream_in, audio_input_flags_t flags __unused, - const char *address __unused, + const char *address, audio_source_t source __unused) { struct submix_audio_device *rsxadev = audio_hw_device_get_submix_audio_device(dev); struct submix_stream_in *in; - ALOGD("adev_open_input_stream()"); + ALOGD("adev_open_input_stream(addr=%s)", address); (void)handle; (void)devices; *stream_in = NULL; + // Do we already have a route for this address + int route_idx = -1; + + pthread_mutex_lock(&rsxadev->lock); + + status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx); + if (res != OK) { + ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address); + pthread_mutex_unlock(&rsxadev->lock); + return res; + } + // Make sure it's possible to open the device given the current audio config. submix_sanitize_config(config, true); - if (!submix_open_validate(rsxadev, &rsxadev->lock, config, true)) { + if (!submix_open_validate_l(rsxadev, route_idx, config, true)) { ALOGE("adev_open_input_stream(): Unable to open input stream."); + pthread_mutex_unlock(&rsxadev->lock); return -EINVAL; } #if ENABLE_LEGACY_INPUT_OPEN - pthread_mutex_lock(&rsxadev->lock); - in = rsxadev->input; + in = rsxadev->routes[route_idx].input; if (in) { in->ref_count++; - sp<MonoPipe> sink = rsxadev->rsxSink; + sp<MonoPipe> sink = rsxadev->routes[route_idx].rsxSink; ALOG_ASSERT(sink != NULL); // If the sink has been shutdown, delete the pipe. if (sink != NULL) { if (sink->isShutdown()) { ALOGD(" Non-NULL shut down sink when opening input stream, releasing, refcount=%d", in->ref_count); - submix_audio_device_release_pipe(rsxadev); + submix_audio_device_release_pipe_l(rsxadev, in->route_handle); } else { ALOGD(" Non-NULL sink when opening input stream, refcount=%d", in->ref_count); } @@ -1420,7 +1546,6 @@ static int adev_open_input_stream(struct audio_hw_device *dev, ALOGE("NULL sink when opening input stream, refcount=%d", in->ref_count); } } - pthread_mutex_unlock(&rsxadev->lock); #else in = NULL; #endif // ENABLE_LEGACY_INPUT_OPEN @@ -1446,18 +1571,29 @@ static int adev_open_input_stream(struct audio_hw_device *dev, in->stream.set_gain = in_set_gain; in->stream.read = in_read; in->stream.get_input_frames_lost = in_get_input_frames_lost; + + in->dev = rsxadev; +#if LOG_STREAMS_TO_FILES + in->log_fd = -1; +#endif } // Initialize the input stream. in->read_counter_frames = 0; - in->output_standby = rsxadev->output_standby; - in->dev = rsxadev; + in->input_standby = true; + if (rsxadev->routes[route_idx].output != NULL) { + in->output_standby_rec_thr = rsxadev->routes[route_idx].output->output_standby; + } else { + in->output_standby_rec_thr = true; + } + in->read_error_count = 0; // Initialize the pipe. ALOGV("adev_open_input_stream(): about to create pipe"); - submix_audio_device_create_pipe(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, - DEFAULT_PIPE_PERIOD_COUNT, in, NULL); + submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, + DEFAULT_PIPE_PERIOD_COUNT, in, NULL, address, route_idx); #if LOG_STREAMS_TO_FILES + if (in->log_fd >= 0) close(in->log_fd); in->log_fd = open(LOG_STREAM_IN_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, LOG_STREAM_FILE_PERMISSIONS); ALOGE_IF(in->log_fd < 0, "adev_open_input_stream(): log file open failed %s", @@ -1467,15 +1603,19 @@ static int adev_open_input_stream(struct audio_hw_device *dev, // Return the input stream. *stream_in = &in->stream; + pthread_mutex_unlock(&rsxadev->lock); return 0; } static void adev_close_input_stream(struct audio_hw_device *dev, struct audio_stream_in *stream) { + struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); + struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); ALOGD("adev_close_input_stream()"); - submix_audio_device_destroy_pipe(audio_hw_device_get_submix_audio_device(dev), in, NULL); + pthread_mutex_lock(&rsxadev->lock); + submix_audio_device_destroy_pipe_l(rsxadev, in, NULL); #if LOG_STREAMS_TO_FILES if (in->log_fd >= 0) close(in->log_fd); #endif // LOG_STREAMS_TO_FILES @@ -1484,12 +1624,26 @@ static void adev_close_input_stream(struct audio_hw_device *dev, #else free(in); #endif // ENABLE_LEGACY_INPUT_OPEN + + pthread_mutex_unlock(&rsxadev->lock); } static int adev_dump(const audio_hw_device_t *device, int fd) { - (void)device; - (void)fd; + const struct submix_audio_device * rsxadev = //audio_hw_device_get_submix_audio_device(device); + reinterpret_cast<const struct submix_audio_device *>( + reinterpret_cast<const uint8_t *>(device) - + offsetof(struct submix_audio_device, device)); + char msg[100]; + int n = sprintf(msg, "\nReroute submix audio module:\n"); + write(fd, &msg, n); + for (int i=0 ; i < MAX_ROUTES ; i++) { + n = sprintf(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); + write(fd, &msg, n); + } return 0; } @@ -1536,8 +1690,10 @@ static int adev_open(const hw_module_t* module, const char* name, rsxadev->device.close_input_stream = adev_close_input_stream; rsxadev->device.dump = adev_dump; - rsxadev->input_standby = true; - rsxadev->output_standby = true; + for (int i=0 ; i < MAX_ROUTES ; i++) { + memset(&rsxadev->routes[i], 0, sizeof(route_config)); + strcpy(rsxadev->routes[i].address, ""); + } *device = &rsxadev->device.common; diff --git a/modules/sensors/multihal.cpp b/modules/sensors/multihal.cpp index 76ec161..cd67f6d 100644 --- a/modules/sensors/multihal.cpp +++ b/modules/sensors/multihal.cpp @@ -28,6 +28,7 @@ #include <vector> #include <map> +#include <string> #include <stdio.h> #include <dlfcn.h> @@ -250,13 +251,41 @@ int sensors_poll_context_t::get_device_version_by_handle(int handle) { } } +// Android L requires sensor HALs to be either 1_0 or 1_3 compliant +#define HAL_VERSION_IS_COMPLIANT(version) \ + (version == SENSORS_DEVICE_API_VERSION_1_0 || version >= SENSORS_DEVICE_API_VERSION_1_3) + +// Returns true if HAL is compliant, false if HAL is not compliant or if handle is invalid +static bool halIsCompliant(sensors_poll_context_t *ctx, int handle) { + int version = ctx->get_device_version_by_handle(handle); + return version != -1 && HAL_VERSION_IS_COMPLIANT(version); +} + +const char *apiNumToStr(int version) { + switch(version) { + case SENSORS_DEVICE_API_VERSION_1_0: + return "SENSORS_DEVICE_API_VERSION_1_0"; + case SENSORS_DEVICE_API_VERSION_1_1: + return "SENSORS_DEVICE_API_VERSION_1_1"; + case SENSORS_DEVICE_API_VERSION_1_2: + return "SENSORS_DEVICE_API_VERSION_1_2"; + case SENSORS_DEVICE_API_VERSION_1_3: + return "SENSORS_DEVICE_API_VERSION_1_3"; + default: + return "UNKNOWN"; + } +} + int sensors_poll_context_t::activate(int handle, int enabled) { int retval = -EINVAL; ALOGV("activate"); int local_handle = get_local_handle(handle); sensors_poll_device_t* v0 = this->get_v0_device_by_handle(handle); - if (local_handle >= 0 && v0) { + if (halIsCompliant(this, handle) && local_handle >= 0 && v0) { retval = v0->activate(v0, local_handle, enabled); + } else { + ALOGE("IGNORING activate(enable %d) call to non-API-compliant sensor handle=%d !", + enabled, handle); } ALOGV("retval %d", retval); return retval; @@ -267,8 +296,10 @@ int sensors_poll_context_t::setDelay(int handle, int64_t ns) { ALOGV("setDelay"); int local_handle = get_local_handle(handle); sensors_poll_device_t* v0 = this->get_v0_device_by_handle(handle); - if (local_handle >= 0 && v0) { + if (halIsCompliant(this, handle) && local_handle >= 0 && v0) { retval = v0->setDelay(v0, local_handle, ns); + } else { + ALOGE("IGNORING setDelay() call for non-API-compliant sensor handle=%d !", handle); } ALOGV("retval %d", retval); return retval; @@ -341,11 +372,12 @@ int sensors_poll_context_t::poll(sensors_event_t *data, int maxReads) { int sensors_poll_context_t::batch(int handle, int flags, int64_t period_ns, int64_t timeout) { ALOGV("batch"); int retval = -EINVAL; - int version = this->get_device_version_by_handle(handle); int local_handle = get_local_handle(handle); sensors_poll_device_1_t* v1 = this->get_v1_device_by_handle(handle); - if (version >= SENSORS_DEVICE_API_VERSION_1_0 && local_handle >= 0 && v1) { + if (halIsCompliant(this, handle) && local_handle >= 0 && v1) { retval = v1->batch(v1, local_handle, flags, period_ns, timeout); + } else { + ALOGE("IGNORING batch() call to non-API-compliant sensor handle=%d !", handle); } ALOGV("retval %d", retval); return retval; @@ -354,11 +386,12 @@ int sensors_poll_context_t::batch(int handle, int flags, int64_t period_ns, int6 int sensors_poll_context_t::flush(int handle) { ALOGV("flush"); int retval = -EINVAL; - int version = this->get_device_version_by_handle(handle); int local_handle = get_local_handle(handle); sensors_poll_device_1_t* v1 = this->get_v1_device_by_handle(handle); - if (version >= SENSORS_DEVICE_API_VERSION_1_0 && local_handle >= 0 && v1) { + if (halIsCompliant(this, handle) && local_handle >= 0 && v1) { retval = v1->flush(v1, local_handle); + } else { + ALOGE("IGNORING flush() call to non-API-compliant sensor handle=%d !", handle); } ALOGV("retval %d", retval); return retval; @@ -577,7 +610,7 @@ static void lazy_init_sensors_list() { ALOGV("end lazy_init_sensors_list"); } -static int module__get_sensors_list(struct sensors_module_t* module, +static int module__get_sensors_list(__unused struct sensors_module_t* module, struct sensor_t const** list) { ALOGV("module__get_sensors_list start"); lazy_init_sensors_list(); @@ -618,7 +651,7 @@ static int open_sensors(const struct hw_module_t* hw_module, const char* name, sensors_poll_context_t *dev = new sensors_poll_context_t(); memset(dev, 0, sizeof(sensors_poll_device_1_t)); dev->proxy_device.common.tag = HARDWARE_DEVICE_TAG; - dev->proxy_device.common.version = SENSORS_DEVICE_API_VERSION_1_1; + dev->proxy_device.common.version = SENSORS_DEVICE_API_VERSION_1_3; dev->proxy_device.common.module = const_cast<hw_module_t*>(hw_module); dev->proxy_device.common.close = device__close; dev->proxy_device.activate = device__activate; @@ -635,8 +668,15 @@ static int open_sensors(const struct hw_module_t* hw_module, const char* name, sensors_module_t *sensors_module = (sensors_module_t*) *it; struct hw_device_t* sub_hw_device; int sub_open_result = sensors_module->common.methods->open(*it, name, &sub_hw_device); - if (!sub_open_result) + if (!sub_open_result) { + if (!HAL_VERSION_IS_COMPLIANT(sub_hw_device->version)) { + ALOGE("SENSORS_DEVICE_API_VERSION_1_3 is required for all sensor HALs"); + ALOGE("This HAL reports non-compliant API level : %s", + apiNumToStr(sub_hw_device->version)); + ALOGE("Sensors belonging to this HAL will get ignored !"); + } dev->addSubHwDevice(sub_hw_device); + } } // Prepare the output param and return diff --git a/modules/usbaudio/alsa_device_profile.c b/modules/usbaudio/alsa_device_profile.c index c7df00c..8e84471 100644 --- a/modules/usbaudio/alsa_device_profile.c +++ b/modules/usbaudio/alsa_device_profile.c @@ -58,10 +58,9 @@ extern int8_t const pcm_format_value_map[50]; static const unsigned std_sample_rates[] = {48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000}; -void profile_init(alsa_device_profile* profile, int direction) +static void profile_reset(alsa_device_profile* profile) { profile->card = profile->device = -1; - profile->direction = direction; /* Fill the attribute arrays with invalid values */ size_t index; @@ -83,6 +82,12 @@ void profile_init(alsa_device_profile* profile, int direction) profile->is_valid = false; } +void profile_init(alsa_device_profile* profile, int direction) +{ + profile->direction = direction; + profile_reset(profile); +} + bool profile_is_initialized(alsa_device_profile* profile) { return profile->card >= 0 && profile->device >= 0; @@ -97,7 +102,7 @@ bool profile_is_cached_for(alsa_device_profile* profile, int card, int device) { } void profile_decache(alsa_device_profile* profile) { - profile->card = profile->device = -1; + profile_reset(profile); } /* @@ -275,28 +280,19 @@ static unsigned profile_enum_sample_formats(alsa_device_profile* profile, struct static unsigned profile_enum_channel_counts(alsa_device_profile* profile, unsigned min, unsigned max) { - // TODO: Don't return MONO even if the device supports it. This causes problems - // in AudioPolicyManager. Revisit. - static const unsigned std_out_channel_counts[] = {8, 4, 2/*, 1*/}; - static const unsigned std_in_channel_counts[] = {8, 4, 2, 1}; - - unsigned * channel_counts = - profile->direction == PCM_OUT ? std_out_channel_counts : std_in_channel_counts; - unsigned num_channel_counts = - profile->direction == PCM_OUT - ? ARRAY_SIZE(std_out_channel_counts) : ARRAY_SIZE(std_in_channel_counts); + static const unsigned std_channel_counts[] = {8, 4, 2, 1}; unsigned num_counts = 0; unsigned index; /* TODO write a profile_test_channel_count() */ /* Ensure there is at least one invalid channel count to terminate the channel counts array */ - for (index = 0; index < num_channel_counts && + for (index = 0; index < ARRAY_SIZE(std_channel_counts) && num_counts < ARRAY_SIZE(profile->channel_counts) - 1; index++) { /* TODO Do we want a channel counts test? */ - if (channel_counts[index] >= min && channel_counts[index] <= max /* && + if (std_channel_counts[index] >= min && std_channel_counts[index] <= max /* && profile_test_channel_count(profile, channel_counts[index])*/) { - profile->channel_counts[num_counts++] = channel_counts[index]; + profile->channel_counts[num_counts++] = std_channel_counts[index]; } } @@ -459,6 +455,7 @@ char * profile_get_channel_count_strs(alsa_device_profile* profile) }; const bool isOutProfile = profile->direction == PCM_OUT; + const char * const * const names_array = isOutProfile ? out_chans_strs : in_chans_strs; const size_t names_size = isOutProfile ? ARRAY_SIZE(out_chans_strs) : ARRAY_SIZE(in_chans_strs); @@ -467,12 +464,17 @@ char * profile_get_channel_count_strs(alsa_device_profile* profile) buffer[0] = '\0'; const int buffer_size = ARRAY_SIZE(buffer); int num_entries = 0; - bool stereo_allowed = false; + /* We currently support MONO and STEREO, and always report STEREO but some (many) + * USB Audio Devices may only announce support for MONO (a headset mic for example), or + * The total number of output channels. SO, if the device itself doesn't explicitly + * support STEREO, append to the channel config strings we are generating. + */ + bool stereo_present = false; unsigned index; unsigned channel_count; for (index = 0; (channel_count = profile->channel_counts[index]) != 0; index++) { - stereo_allowed = stereo_allowed || channel_count == 2; + stereo_present = stereo_present || channel_count == 2; if (channel_count < names_size && names_array[channel_count] != NULL) { if (num_entries++ != 0) { strncat(buffer, "|", buffer_size); @@ -480,14 +482,16 @@ char * profile_get_channel_count_strs(alsa_device_profile* profile) strncat(buffer, names_array[channel_count], buffer_size); } } + /* emulated modes: * always expose stereo as we can emulate it for PCM_OUT */ - if (!stereo_allowed && isOutProfile) { + if (!stereo_present) { if (num_entries++ != 0) { strncat(buffer, "|", buffer_size); } strncat(buffer, names_array[2], buffer_size); /* stereo */ } + return strdup(buffer); } diff --git a/modules/usbaudio/audio_hw.c b/modules/usbaudio/audio_hw.c index 78fa7c5..ad01833 100644 --- a/modules/usbaudio/audio_hw.c +++ b/modules/usbaudio/audio_hw.c @@ -104,9 +104,11 @@ struct stream_in { alsa_device_profile * profile; alsa_device_proxy proxy; /* state of the stream */ - // not used? - // struct audio_config hal_pcm_config; - + unsigned hal_channel_count; /* channel count exposed to AudioFlinger. + * This may differ from the device channel count when + * the device is not compatible with AudioFlinger + * capabilities, e.g. exposes too many channels or + * too few channels. */ /* We may need to read more data from the device in order to data reduce to 16bit, 4chan */ void * conversion_buffer; /* any conversions are put into here * they could come from here too if @@ -623,25 +625,13 @@ static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) static size_t in_get_buffer_size(const struct audio_stream *stream) { const struct stream_in * in = ((const struct stream_in*)stream); - size_t buffer_size = - proxy_get_period_size(&in->proxy) * audio_stream_in_frame_size(&(in->stream)); - ALOGV("in_get_buffer_size() = %zd", buffer_size); - - return buffer_size; + return proxy_get_period_size(&in->proxy) * audio_stream_in_frame_size(&(in->stream)); } static uint32_t in_get_channels(const struct audio_stream *stream) { - /* TODO Here is the code we need when we support arbitrary channel counts - * alsa_device_proxy * proxy = ((struct stream_in*)stream)->proxy; - * unsigned channel_count = proxy_get_channel_count(proxy); - * uint32_t channel_mask = audio_channel_in_mask_from_count(channel_count); - * ALOGV("in_get_channels() = 0x%X count:%d", channel_mask, channel_count); - * return channel_mask; - */ - /* TODO When AudioPolicyManager & AudioFlinger supports arbitrary channels - rewrite this to return the ACTUAL channel format */ - return AUDIO_CHANNEL_IN_STEREO; + const struct stream_in *in = (const struct stream_in*)stream; + return audio_channel_in_mask_from_count(in->hal_channel_count); } static audio_format_t in_get_format(const struct audio_stream *stream) @@ -785,6 +775,7 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t byte size_t num_read_buff_bytes = 0; void * read_buff = buffer; void * out_buff = buffer; + int ret = 0; struct stream_in * in = (struct stream_in *)stream; @@ -808,7 +799,7 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t byte */ num_read_buff_bytes = bytes; int num_device_channels = proxy_get_channel_count(&in->proxy); - int num_req_channels = 2; /* always, for now */ + int num_req_channels = in->hal_channel_count; if (num_device_channels != num_req_channels) { num_read_buff_bytes = (num_device_channels * num_read_buff_bytes) / num_req_channels; @@ -834,7 +825,8 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t byte read_buff = in->conversion_buffer; } - if (proxy_read(&in->proxy, read_buff, num_read_buff_bytes) == 0) { + ret = proxy_read(&in->proxy, read_buff, num_read_buff_bytes); + if (ret == 0) { /* * Do any conversions necessary to send the data in the format specified to/by the HAL * (but different from the ALSA format), such as 24bit ->16bit, or 4chan -> 2chan. @@ -878,7 +870,7 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t byte if (num_read_buff_bytes > 0 && in->dev->mic_muted) memset(buffer, 0, num_read_buff_bytes); } else { - num_read_buff_bytes = 0; + num_read_buff_bytes = 0; // reset the value after USB headset is unplugged } err: @@ -964,19 +956,18 @@ static int adev_open_input_stream(struct audio_hw_device *dev, ret = -EINVAL; } - if (config->channel_mask == AUDIO_CHANNEL_NONE) { - /* just return AUDIO_CHANNEL_IN_STEREO until the framework supports other input - * formats */ - config->channel_mask = AUDIO_CHANNEL_IN_STEREO; - - } else if (config->channel_mask != AUDIO_CHANNEL_IN_STEREO) { - /* allow only stereo capture for now */ - config->channel_mask = AUDIO_CHANNEL_IN_STEREO; - ret = -EINVAL; + /* Channels */ + unsigned proposed_channel_count = profile_get_default_channel_count(in->profile); + if (k_force_channels) { + proposed_channel_count = k_force_channels; + } else if (config->channel_mask != AUDIO_CHANNEL_NONE) { + proposed_channel_count = audio_channel_count_from_in_mask(config->channel_mask); } - // proxy_config.channels = 0; /* don't change */ - proxy_config.channels = profile_get_default_channel_count(in->profile); + /* we can expose any channel count mask, and emulate internally. */ + config->channel_mask = audio_channel_in_mask_from_count(proposed_channel_count); + in->hal_channel_count = proposed_channel_count; + proxy_config.channels = profile_get_default_channel_count(in->profile); proxy_prepare(&in->proxy, in->profile, &proxy_config); in->standby = true; |