зеркало из https://github.com/mozilla/gecko-dev.git
Bug 1410456 - remove preferred sample rate implementation because makes use of dlopen. r=padenot
MozReview-Commit-ID: jBZ3oewbjh --HG-- extra : rebase_source : e983510c4c3e2e046302ea3c0732b25a0cfd2cb1
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cb6847750a
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9dba55fc1f
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@ -154,7 +154,7 @@ struct cubeb_stream {
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cubeb_state_callback state_callback;
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cubeb_resampler * resampler;
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unsigned int inputrate;
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unsigned int user_output_rate;
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unsigned int output_configured_rate;
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unsigned int latency_frames;
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int64_t lastPosition;
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@ -784,66 +784,6 @@ opensl_get_max_channel_count(cubeb * ctx, uint32_t * max_channels)
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return CUBEB_OK;
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}
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static int
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opensl_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate)
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{
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/* https://android.googlesource.com/platform/ndk.git/+/master/docs/opensles/index.html
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* We don't want to deal with JNI here (and we don't have Java on b2g anyways),
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* so we just dlopen the library and get the two symbols we need. */
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int r;
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void * libmedia;
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uint32_t (*get_primary_output_samplingrate)();
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uint32_t (*get_output_samplingrate)(int * samplingRate, int streamType);
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libmedia = dlopen("libmedia.so", RTLD_LAZY);
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if (!libmedia) {
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return CUBEB_ERROR;
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}
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/* uint32_t AudioSystem::getPrimaryOutputSamplingRate(void) */
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get_primary_output_samplingrate =
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dlsym(libmedia, "_ZN7android11AudioSystem28getPrimaryOutputSamplingRateEv");
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if (!get_primary_output_samplingrate) {
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/* fallback to
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* status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType)
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* if we cannot find getPrimaryOutputSamplingRate. */
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get_output_samplingrate =
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dlsym(libmedia, "_ZN7android11AudioSystem21getOutputSamplingRateEPj19audio_stream_type_t");
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if (!get_output_samplingrate) {
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/* Another signature exists, with a int instead of an audio_stream_type_t */
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get_output_samplingrate =
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dlsym(libmedia, "_ZN7android11AudioSystem21getOutputSamplingRateEPii");
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if (!get_output_samplingrate) {
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dlclose(libmedia);
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return CUBEB_ERROR;
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}
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}
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}
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if (get_primary_output_samplingrate) {
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*rate = get_primary_output_samplingrate();
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} else {
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/* We don't really know about the type, here, so we just pass music. */
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r = get_output_samplingrate((int *) rate, AUDIO_STREAM_TYPE_MUSIC);
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if (r) {
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dlclose(libmedia);
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return CUBEB_ERROR;
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}
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}
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dlclose(libmedia);
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/* Depending on which method we called above, we can get a zero back, yet have
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* a non-error return value, especially if the audio system is not
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* ready/shutting down (i.e. when we can't get our hand on the AudioFlinger
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* thread). */
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if (*rate == 0) {
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return CUBEB_ERROR;
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}
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return CUBEB_OK;
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}
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static void
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opensl_destroy(cubeb * ctx)
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{
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@ -939,13 +879,9 @@ opensl_configure_capture(cubeb_stream * stm, cubeb_stream_params * params)
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// api for input device this is a safe choice.
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stm->input_device_rate = stm->output_configured_rate;
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} else {
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// The output preferred rate is used for input only scenario. This is
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// the correct rate to use to get a fast track for input only.
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r = opensl_get_preferred_sample_rate(stm->context, &stm->input_device_rate);
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if (r != CUBEB_OK) {
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// If everything else fail use a safe choice for Android.
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stm->input_device_rate = DEFAULT_SAMPLE_RATE;
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}
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// The output preferred rate is used for input only scenario. The
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// default rate expected to supported from all android devices.
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stm->input_device_rate = DEFAULT_SAMPLE_RATE;
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}
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lDataFormat.samplesPerSec = stm->input_device_rate * 1000;
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res = (*stm->context->eng)->CreateAudioRecorder(stm->context->eng,
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@ -1056,7 +992,7 @@ opensl_configure_playback(cubeb_stream * stm, cubeb_stream_params * params) {
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assert(stm);
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assert(params);
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stm->inputrate = params->rate;
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stm->user_output_rate = params->rate;
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stm->framesize = params->channels * sizeof(int16_t);
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stm->lastPosition = -1;
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stm->lastPositionTimeStamp = 0;
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@ -1093,7 +1029,7 @@ opensl_configure_playback(cubeb_stream * stm, cubeb_stream_params * params) {
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#endif
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assert(NELEMS(ids) == NELEMS(req));
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uint32_t preferred_sampling_rate = stm->inputrate;
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uint32_t preferred_sampling_rate = stm->user_output_rate;
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SLresult res = SL_RESULT_CONTENT_UNSUPPORTED;
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if (preferred_sampling_rate) {
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res = (*stm->context->eng)->CreateAudioPlayer(stm->context->eng,
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@ -1106,12 +1042,9 @@ opensl_configure_playback(cubeb_stream * stm, cubeb_stream_params * params) {
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}
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// Sample rate not supported? Try again with primary sample rate!
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if (res == SL_RESULT_CONTENT_UNSUPPORTED) {
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if (opensl_get_preferred_sample_rate(stm->context, &preferred_sampling_rate)) {
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// If fail default is used
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preferred_sampling_rate = DEFAULT_SAMPLE_RATE;
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}
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if (res == SL_RESULT_CONTENT_UNSUPPORTED &&
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preferred_sampling_rate != DEFAULT_SAMPLE_RATE) {
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preferred_sampling_rate = DEFAULT_SAMPLE_RATE;
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format.samplesPerSec = preferred_sampling_rate * 1000;
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res = (*stm->context->eng)->CreateAudioPlayer(stm->context->eng,
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&stm->playerObj,
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@ -1550,7 +1483,7 @@ opensl_stream_get_position(cubeb_stream * stm, uint64_t * position)
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stm->lastPosition = msec;
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}
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samplerate = stm->inputrate;
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samplerate = stm->user_output_rate;
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r = stm->context->get_output_latency(&mixer_latency, AUDIO_STREAM_TYPE_MUSIC);
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if (r) {
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@ -1558,7 +1491,7 @@ opensl_stream_get_position(cubeb_stream * stm, uint64_t * position)
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}
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pthread_mutex_lock(&stm->mutex);
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int64_t maximum_position = stm->written * (int64_t)stm->inputrate / stm->output_configured_rate;
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int64_t maximum_position = stm->written * (int64_t)stm->user_output_rate / stm->output_configured_rate;
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pthread_mutex_unlock(&stm->mutex);
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assert(maximum_position >= 0);
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@ -1617,7 +1550,7 @@ static struct cubeb_ops const opensl_ops = {
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.get_backend_id = opensl_get_backend_id,
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.get_max_channel_count = opensl_get_max_channel_count,
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.get_min_latency = NULL,
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.get_preferred_sample_rate = opensl_get_preferred_sample_rate,
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.get_preferred_sample_rate = NULL,
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.get_preferred_channel_layout = NULL,
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.enumerate_devices = NULL,
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.device_collection_destroy = NULL,
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