Decouple input and output sample rate overrides.

We may sometimes want to override only input or only output, or
override them with different values. This change allows setting the
overrides separately.


Change-Id: Ib0c44cb7a3cfa834f997fb6cd54f7cad68705f41
Bug: webrtc:10441
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128763
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27236}
This commit is contained in:
Paulina Hensman
2019-03-21 16:02:41 +01:00
committed by Commit Bot
parent 2293622f02
commit 1ca30a7e41
6 changed files with 61 additions and 28 deletions

View File

@ -615,7 +615,8 @@ int GetDefaultSampleRate(JNIEnv* env, const JavaRef<jobject>& j_audio_manager) {
void GetAudioParameters(JNIEnv* env,
const JavaRef<jobject>& j_context,
const JavaRef<jobject>& j_audio_manager,
int sample_rate,
int input_sample_rate,
int output_sample_rate,
bool use_stereo_input,
bool use_stereo_output,
AudioParameters* input_parameters,
@ -623,12 +624,14 @@ void GetAudioParameters(JNIEnv* env,
const int output_channels = use_stereo_output ? 2 : 1;
const int input_channels = use_stereo_input ? 2 : 1;
const size_t output_buffer_size = Java_WebRtcAudioManager_getOutputBufferSize(
env, j_context, j_audio_manager, sample_rate, output_channels);
env, j_context, j_audio_manager, output_sample_rate, output_channels);
const size_t input_buffer_size = Java_WebRtcAudioManager_getInputBufferSize(
env, j_context, j_audio_manager, sample_rate, input_channels);
output_parameters->reset(sample_rate, static_cast<size_t>(output_channels),
env, j_context, j_audio_manager, input_sample_rate, input_channels);
output_parameters->reset(output_sample_rate,
static_cast<size_t>(output_channels),
static_cast<size_t>(output_buffer_size));
input_parameters->reset(sample_rate, static_cast<size_t>(input_channels),
input_parameters->reset(input_sample_rate,
static_cast<size_t>(input_channels),
static_cast<size_t>(input_buffer_size));
RTC_CHECK(input_parameters->is_valid());
RTC_CHECK(output_parameters->is_valid());

View File

@ -78,7 +78,8 @@ int GetDefaultSampleRate(JNIEnv* env, const JavaRef<jobject>& j_audio_manager);
void GetAudioParameters(JNIEnv* env,
const JavaRef<jobject>& j_context,
const JavaRef<jobject>& j_audio_manager,
int sample_rate,
int input_sample_rate,
int output_sample_rate,
bool use_stereo_input,
bool use_stereo_output,
AudioParameters* input_parameters,

View File

@ -23,13 +23,15 @@ static jlong JNI_JavaAudioDeviceModule_CreateAudioDeviceModule(
const JavaParamRef<jobject>& j_audio_manager,
const JavaParamRef<jobject>& j_webrtc_audio_record,
const JavaParamRef<jobject>& j_webrtc_audio_track,
int sample_rate,
int input_sample_rate,
int output_sample_rate,
jboolean j_use_stereo_input,
jboolean j_use_stereo_output) {
AudioParameters input_parameters;
AudioParameters output_parameters;
GetAudioParameters(env, j_context, j_audio_manager, sample_rate,
j_use_stereo_input, j_use_stereo_output, &input_parameters,
GetAudioParameters(env, j_context, j_audio_manager, input_sample_rate,
output_sample_rate, j_use_stereo_input,
j_use_stereo_output, &input_parameters,
&output_parameters);
auto audio_input = absl::make_unique<AudioRecordJni>(
env, input_parameters, kHighLatencyModeDelayEstimateInMilliseconds,