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Fix for glitches in ACM when switching desired output sample rate

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The problem was that if the output sample rate is changed such from one
where no resampling is needed to a rate that requires resampling, the
first output from the resampler will contain an onset period. The
solution provided in this CL is to keep a copy of the last output frame
in ACM, and if the resampler is engaged, it will be primed with this
old frame before resampling the current frame.

BUG=3919
R=bjornv@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7479 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
2014-10-21 06:54:23 +00:00
parent a8c0edd29f
commit 913f7b8d5e
3 changed files with 62 additions and 57 deletions
Download Patch File Download Diff File Expand all files Collapse all files

111
webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
View File

@@ -122,11 +122,14 @@ AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
last_audio_decoder_(-1), // Invalid value.
previous_audio_activity_(AudioFrame::kVadPassive),
current_sample_rate_hz_(config.neteq_config.sample_rate_hz),
audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
nack_(),
nack_enabled_(false),
neteq_(NetEq::Create(config.neteq_config)),
vad_enabled_(true),
clock_(config.clock),
resampled_last_output_frame_(true),
av_sync_(false),
initial_delay_manager_(),
missing_packets_sync_stream_(),
@@ -143,6 +146,9 @@ AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
neteq_->EnableVad();
else
neteq_->DisableVad();
memset(audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
}
AcmReceiver::~AcmReceiver() {
@@ -342,7 +348,6 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
enum NetEqOutputType type;
int16_t* ptr_audio_buffer = audio_frame->data_;
int samples_per_channel;
int num_channels;
bool return_silence = false;
@@ -359,18 +364,6 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
initial_delay_manager_->LatePackets(timestamp_now,
late_packets_sync_stream_.get());
}
if (!return_silence) {
// This is our initial guess regarding whether a resampling will be
// required. It is based on previous sample rate of netEq. Most often,
// this is a correct guess, however, in case that incoming payload changes
// the resampling might might be needed. By doing so, we avoid an
// unnecessary memcpy().
if (desired_freq_hz != -1 &&
current_sample_rate_hz_ != desired_freq_hz) {
ptr_audio_buffer = audio_buffer_;
}
}
}
// If |late_packets_sync_stream_| is allocated then we have been in AV-sync
@@ -381,17 +374,19 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
return 0;
}
// Accessing members, take the lock.
CriticalSectionScoped lock(crit_sect_.get());
// Always write the output to |audio_buffer_| first.
if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
ptr_audio_buffer,
audio_buffer_.get(),
&samples_per_channel,
&num_channels, &type) != NetEq::kOK) {
&num_channels,
&type) != NetEq::kOK) {
LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "NetEq Failed.";
return -1;
}
// Accessing members, take the lock.
CriticalSectionScoped lock(crit_sect_.get());
// Update NACK.
int decoded_sequence_num = 0;
uint32_t decoded_timestamp = 0;
@@ -409,45 +404,53 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
bool need_resampling = (desired_freq_hz != -1) &&
(current_sample_rate_hz_ != desired_freq_hz);
if (ptr_audio_buffer == audio_buffer_) {
// Data is written to local buffer.
if (need_resampling) {
samples_per_channel =
resampler_.Resample10Msec(audio_buffer_,
current_sample_rate_hz_,
desired_freq_hz,
num_channels,
AudioFrame::kMaxDataSizeSamples,
audio_frame->data_);
if (samples_per_channel < 0) {
LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "Resampler Failed.";
return -1;
}
} else {
// We might end up here ONLY if codec is changed.
memcpy(audio_frame->data_, audio_buffer_, samples_per_channel *
num_channels * sizeof(int16_t));
}
} else {
// Data is written into |audio_frame|.
if (need_resampling) {
// We might end up here ONLY if codec is changed.
samples_per_channel =
resampler_.Resample10Msec(audio_frame->data_,
current_sample_rate_hz_,
desired_freq_hz,
num_channels,
AudioFrame::kMaxDataSizeSamples,
audio_buffer_);
if (samples_per_channel < 0) {
LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "Resampler Failed.";
return -1;
}
memcpy(audio_frame->data_, audio_buffer_, samples_per_channel *
num_channels * sizeof(int16_t));
if (need_resampling && !resampled_last_output_frame_) {
// Prime the resampler with the last frame.
int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
samples_per_channel =
resampler_.Resample10Msec(last_audio_buffer_.get(),
current_sample_rate_hz_,
desired_freq_hz,
num_channels,
AudioFrame::kMaxDataSizeSamples,
temp_output);
if (samples_per_channel < 0) {
LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio")
<< "Resampling last_audio_buffer_ failed.";
return -1;
}
}
// The audio in |audio_buffer_| is tansferred to |audio_frame_| below, either
// through resampling, or through straight memcpy.
// TODO(henrik.lundin) Glitches in the output may appear if the output rate
// from NetEq changes. See WebRTC issue 3923.
if (need_resampling) {
samples_per_channel =
resampler_.Resample10Msec(audio_buffer_.get(),
current_sample_rate_hz_,
desired_freq_hz,
num_channels,
AudioFrame::kMaxDataSizeSamples,
audio_frame->data_);
if (samples_per_channel < 0) {
LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio")
<< "Resampling audio_buffer_ failed.";
return -1;
}
resampled_last_output_frame_ = true;
} else {
resampled_last_output_frame_ = false;
// We might end up here ONLY if codec is changed.
memcpy(audio_frame->data_,
audio_buffer_.get(),
samples_per_channel * num_channels * sizeof(int16_t));
}
// Swap buffers, so that the current audio is stored in |last_audio_buffer_|
// for next time.
audio_buffer_.swap(last_audio_buffer_);
audio_frame->num_channels_ = num_channels;
audio_frame->samples_per_channel_ = samples_per_channel;
audio_frame->sample_rate_hz_ = samples_per_channel * 100;

4
webrtc/modules/audio_coding/main/acm2/acm_receiver.h
View File

@@ -334,7 +334,8 @@ class AcmReceiver {
ACMResampler resampler_ GUARDED_BY(crit_sect_);
// Used in GetAudio, declared as member to avoid allocating every 10ms.
// TODO(henrik.lundin) Stack-allocate in GetAudio instead?
int16_t audio_buffer_[AudioFrame::kMaxDataSizeSamples] GUARDED_BY(crit_sect_);
scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_);
scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
scoped_ptr<Nack> nack_ GUARDED_BY(crit_sect_);
bool nack_enabled_ GUARDED_BY(crit_sect_);
CallStatistics call_stats_ GUARDED_BY(crit_sect_);
@@ -342,6 +343,7 @@ class AcmReceiver {
Decoder decoders_[ACMCodecDB::kMaxNumCodecs];
bool vad_enabled_;
Clock* clock_; // TODO(henrik.lundin) Make const if possible.
bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
// Indicates if a non-zero initial delay is set, and the receiver is in
// AV-sync mode.

4
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
View File

@@ -1034,7 +1034,7 @@ TEST_F(AcmSwitchingOutputFrequencyOldApi, TestWithoutToggling) {
Run(16000, 16000, 1000);
}
TEST_F(AcmSwitchingOutputFrequencyOldApi, DISABLED_Toggle16KhzTo32Khz) {
TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle16KhzTo32Khz) {
Run(16000, 32000, 1000);
}
@@ -1042,7 +1042,7 @@ TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle32KhzTo16Khz) {
Run(32000, 16000, 1000);
}
TEST_F(AcmSwitchingOutputFrequencyOldApi, DISABLED_Toggle16KhzTo8Khz) {
TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle16KhzTo8Khz) {
Run(16000, 8000, 1000);
}