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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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37
webrtc/modules/audio_device/audio_device_buffer.cc
View File

@ -61,6 +61,10 @@ AudioDeviceBuffer::AudioDeviceBuffer()
last_play_samples_(0),
last_log_stat_time_(0) {
LOG(INFO) << "AudioDeviceBuffer::ctor";
// TODO(henrika): improve buffer handling and ensure that we don't allocate
// more than what is required.
play_buffer_.reset(new int8_t[kMaxBufferSizeBytes]);
rec_buffer_.reset(new int8_t[kMaxBufferSizeBytes]);
}
AudioDeviceBuffer::~AudioDeviceBuffer() {
@ -235,8 +239,7 @@ int32_t AudioDeviceBuffer::StopOutputFileRecording() {
int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
size_t num_samples) {
AllocateRecordingBufferIfNeeded();
RTC_CHECK(rec_buffer_);
UpdateRecordingParameters();
// WebRTC can only receive audio in 10ms chunks, hence we fail if the native
// audio layer tries to deliver something else.
RTC_CHECK_EQ(num_samples, rec_samples_per_10ms_);
@ -269,7 +272,6 @@ int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
}
int32_t AudioDeviceBuffer::DeliverRecordedData() {
RTC_CHECK(rec_buffer_);
RTC_DCHECK(audio_transport_cb_);
rtc::CritScope lock(&_critSectCb);
@ -306,8 +308,7 @@ int32_t AudioDeviceBuffer::RequestPlayoutData(size_t num_samples) {
last_playout_time_ = now_time;
playout_diff_times_[diff_time]++;
AllocatePlayoutBufferIfNeeded();
RTC_CHECK(play_buffer_);
UpdatePlayoutParameters();
// WebRTC can only provide audio in 10ms chunks, hence we fail if the native
// audio layer asks for something else.
RTC_CHECK_EQ(num_samples, play_samples_per_10ms_);
@ -346,36 +347,22 @@ int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) {
return static_cast<int32_t>(play_samples_per_10ms_);
}
void AudioDeviceBuffer::AllocatePlayoutBufferIfNeeded() {
void AudioDeviceBuffer::UpdatePlayoutParameters() {
RTC_CHECK(play_bytes_per_sample_);
if (play_buffer_)
return;
LOG(INFO) << __FUNCTION__;
rtc::CritScope lock(&_critSect);
// Derive the required buffer size given sample rate and number of channels.
// Update the required buffer size given sample rate and number of channels.
play_samples_per_10ms_ = static_cast<size_t>(play_sample_rate_ * 10 / 1000);
play_bytes_per_10ms_ = play_bytes_per_sample_ * play_samples_per_10ms_;
LOG(INFO) << "playout samples per 10ms: " << play_samples_per_10ms_;
LOG(INFO) << "playout bytes per 10ms: " << play_bytes_per_10ms_;
// Allocate memory for the playout audio buffer. It will always contain audio
// samples corresponding to 10ms of audio to be played out.
play_buffer_.reset(new int8_t[play_bytes_per_10ms_]);
RTC_DCHECK_LE(play_bytes_per_10ms_, kMaxBufferSizeBytes);
}
void AudioDeviceBuffer::AllocateRecordingBufferIfNeeded() {
void AudioDeviceBuffer::UpdateRecordingParameters() {
RTC_CHECK(rec_bytes_per_sample_);
if (rec_buffer_)
return;
LOG(INFO) << __FUNCTION__;
rtc::CritScope lock(&_critSect);
// Derive the required buffer size given sample rate and number of channels.
// Update the required buffer size given sample rate and number of channels.
rec_samples_per_10ms_ = static_cast<size_t>(rec_sample_rate_ * 10 / 1000);
rec_bytes_per_10ms_ = rec_bytes_per_sample_ * rec_samples_per_10ms_;
LOG(INFO) << "recorded samples per 10ms: " << rec_samples_per_10ms_;
LOG(INFO) << "recorded bytes per 10ms: " << rec_bytes_per_10ms_;
// Allocate memory for the recording audio buffer. It will always contain
// audio samples corresponding to 10ms of audio.
rec_buffer_.reset(new int8_t[rec_bytes_per_10ms_]);
RTC_DCHECK_LE(rec_bytes_per_10ms_, kMaxBufferSizeBytes);
}
void AudioDeviceBuffer::StartTimer() {

19
webrtc/modules/audio_device/audio_device_buffer.h
View File

@ -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.