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Avoids crash at device switch on iOS by ensuring that audio parameters can be updated on the fly driven by e.g. switching audio device.

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R=henrikg@webrtc.org

Review URL: https://codereview.webrtc.org/2324113002 .

Cr-Commit-Position: refs/heads/master@{#14158}
This commit is contained in:
henrika
2016-09-09 13:15:37 +02:00
parent 2b11fd2e4b
commit 073378e79a
2 changed files with 23 additions and 33 deletions
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19
webrtc/modules/audio_device/audio_device_buffer.h
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@ -73,8 +73,11 @@ class AudioDeviceBuffer {
int32_t SetTypingStatus(bool typing_status);
private:
void AllocatePlayoutBufferIfNeeded();
void AllocateRecordingBufferIfNeeded();
// Playout and recording parameters can change on the fly. e.g. at device
// switch. These methods ensures that the callback methods always use the
// latest parameters.
void UpdatePlayoutParameters();
void UpdateRecordingParameters();
// Posts the first delayed task in the task queue and starts the periodic
// timer.
@ -131,14 +134,14 @@ class AudioDeviceBuffer {
size_t play_samples_per_10ms_;
size_t play_bytes_per_10ms_;
// Buffer used for recorded audio samples. Size is given by
// |rec_bytes_per_10ms_| and the buffer is allocated in InitRecording() on the
// main/creating thread.
// Buffer used for recorded audio samples. Size is currently fixed
// but it should be changed to be dynamic and correspond to
// |play_bytes_per_10ms_|. TODO(henrika): avoid using fixed (max) size.
std::unique_ptr<int8_t[]> rec_buffer_;
// Buffer used for audio samples to be played out. Size is given by
// |play_bytes_per_10ms_| and the buffer is allocated in InitPlayout() on the
// main/creating thread.
// Buffer used for audio samples to be played out. Size is currently fixed
// but it should be changed to be dynamic and correspond to
// |play_bytes_per_10ms_|. TODO(henrika): avoid using fixed (max) size.
std::unique_ptr<int8_t[]> play_buffer_;
// AGC parameters.