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Make Opus PLC always output 10ms audio.

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BUG: b/143582588
Change-Id: I41ad5f4f91d9af3f595666a8f32b7ab5382605bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158672
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29733}
This commit is contained in:
Minyue Li
2019-11-04 14:47:52 +01:00
committed by Commit Bot
parent 0696eecbd0
commit 8e83c7ac09
7 changed files with 116 additions and 45 deletions
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73
modules/audio_coding/codecs/opus/opus_interface.cc
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@ -11,6 +11,7 @@
#include "modules/audio_coding/codecs/opus/opus_interface.h"
#include "rtc_base/checks.h"
#include "system_wrappers/include/field_trial.h"
enum {
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
@ -25,8 +26,14 @@ enum {
* side, we must allow for packets of that size. NetEq is currently limited
* to 60 ms on the receive side. */
kWebRtcOpusMaxDecodeFrameSizeMs = 120,
// Duration of audio that each call to packet loss concealment covers.
kWebRtcOpusPlcFrameSizeMs = 10,
};
constexpr char kPlcUsePrevDecodedSamplesFieldTrial[] =
"WebRTC-Audio-OpusPlcUsePrevDecodedSamples";
static int FrameSizePerChannel(int frame_size_ms, int sample_rate_hz) {
RTC_DCHECK_GT(frame_size_ms, 0);
RTC_DCHECK_EQ(frame_size_ms % 10, 0);
@ -381,9 +388,14 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst,
if (error == OPUS_OK && state->decoder) {
// Creation of memory all ok.
state->channels = channels;
state->prev_decoded_samples = DefaultFrameSizePerChannel(sample_rate_hz);
state->in_dtx_mode = 0;
state->sample_rate_hz = sample_rate_hz;
state->plc_use_prev_decoded_samples =
webrtc::field_trial::IsEnabled(kPlcUsePrevDecodedSamplesFieldTrial);
if (state->plc_use_prev_decoded_samples) {
state->prev_decoded_samples =
DefaultFrameSizePerChannel(state->sample_rate_hz);
}
state->in_dtx_mode = 0;
*inst = state;
return 0;
}
@ -420,9 +432,14 @@ int16_t WebRtcOpus_MultistreamDecoderCreate(
if (error == OPUS_OK && state->multistream_decoder) {
// Creation of memory all ok.
state->channels = channels;
state->prev_decoded_samples = DefaultFrameSizePerChannel(48000);
state->in_dtx_mode = 0;
state->sample_rate_hz = 48000;
state->plc_use_prev_decoded_samples =
webrtc::field_trial::IsEnabled(kPlcUsePrevDecodedSamplesFieldTrial);
if (state->plc_use_prev_decoded_samples) {
state->prev_decoded_samples =
DefaultFrameSizePerChannel(state->sample_rate_hz);
}
state->in_dtx_mode = 0;
*inst = state;
return 0;
}
@ -517,17 +534,20 @@ static int DecodeNative(OpusDecInst* inst,
static int DecodePlc(OpusDecInst* inst, int16_t* decoded) {
int16_t audio_type = 0;
int decoded_samples;
int plc_samples;
int plc_samples =
FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz);
/* The number of samples we ask for is |number_of_lost_frames| times
* |prev_decoded_samples_|. Limit the number of samples to maximum
* |MaxFrameSizePerChannel()|. */
plc_samples = inst->prev_decoded_samples;
const int max_samples_per_channel =
MaxFrameSizePerChannel(inst->sample_rate_hz);
plc_samples = plc_samples <= max_samples_per_channel
? plc_samples
: max_samples_per_channel;
if (inst->plc_use_prev_decoded_samples) {
/* The number of samples we ask for is |number_of_lost_frames| times
* |prev_decoded_samples_|. Limit the number of samples to maximum
* |MaxFrameSizePerChannel()|. */
plc_samples = inst->prev_decoded_samples;
const int max_samples_per_channel =
MaxFrameSizePerChannel(inst->sample_rate_hz);
plc_samples = plc_samples <= max_samples_per_channel
? plc_samples
: max_samples_per_channel;
}
decoded_samples =
DecodeNative(inst, NULL, 0, plc_samples, decoded, &audio_type, 0);
if (decoded_samples < 0) {
@ -556,8 +576,10 @@ int WebRtcOpus_Decode(OpusDecInst* inst,
return -1;
}
/* Update decoded sample memory, to be used by the PLC in case of losses. */
inst->prev_decoded_samples = decoded_samples;
if (inst->plc_use_prev_decoded_samples) {
/* Update decoded sample memory, to be used by the PLC in case of losses. */
inst->prev_decoded_samples = decoded_samples;
}
return decoded_samples;
}
@ -612,14 +634,17 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst,
}
int WebRtcOpus_PlcDuration(OpusDecInst* inst) {
/* The number of samples we ask for is |number_of_lost_frames| times
* |prev_decoded_samples_|. Limit the number of samples to maximum
* |MaxFrameSizePerChannel()|. */
const int plc_samples = inst->prev_decoded_samples;
const int max_samples_per_channel =
MaxFrameSizePerChannel(inst->sample_rate_hz);
return plc_samples <= max_samples_per_channel ? plc_samples
: max_samples_per_channel;
if (inst->plc_use_prev_decoded_samples) {
/* The number of samples we ask for is |number_of_lost_frames| times
* |prev_decoded_samples_|. Limit the number of samples to maximum
* |MaxFrameSizePerChannel()|. */
const int plc_samples = inst->prev_decoded_samples;
const int max_samples_per_channel =
MaxFrameSizePerChannel(inst->sample_rate_hz);
return plc_samples <= max_samples_per_channel ? plc_samples
: max_samples_per_channel;
}
return FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz);
}
int WebRtcOpus_FecDurationEst(const uint8_t* payload,