zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Reland "Prevent Opus DTX from generating intermittent noise during silence"

Browse Source 
The original CL is reviewed at
https://codereview.webrtc.org/1415173005/

A silly mistake was made at the last patch set, and the CL was reverted. This CL is to fix and reland it.

BUG=

Review URL: https://codereview.webrtc.org/1422213003

Cr-Commit-Position: refs/heads/master@{#10574}
This commit is contained in:
minyue
2015-11-10 03:49:26 -08:00
committed by Commit bot
parent 626252fa66
commit 3cea256806
6 changed files with 377 additions and 62 deletions
Download Patch File Download Diff File Expand all files Collapse all files

105
webrtc/modules/audio_coding/codecs/opus/opus_interface.c
View File

@ -11,6 +11,7 @@
#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
#include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
#include <assert.h>
#include <stdlib.h>
#include <string.h>
@ -29,48 +30,61 @@ enum {
/* Default frame size, 20 ms @ 48 kHz, in samples (for one channel). */
kWebRtcOpusDefaultFrameSize = 960,
// Maximum number of consecutive zeros, beyond or equal to which DTX can fail.
kZeroBreakCount = 157,
#if defined(OPUS_FIXED_POINT)
kZeroBreakValue = 10,
#else
kZeroBreakValue = 1,
#endif
};
int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
int32_t channels,
int32_t application) {
OpusEncInst* state;
if (inst != NULL) {
state = (OpusEncInst*) calloc(1, sizeof(OpusEncInst));
if (state) {
int opus_app;
switch (application) {
case 0: {
opus_app = OPUS_APPLICATION_VOIP;
break;
}
case 1: {
opus_app = OPUS_APPLICATION_AUDIO;
break;
}
default: {
free(state);
return -1;
}
}
int opus_app;
if (!inst)
return -1;
int error;
state->encoder = opus_encoder_create(48000, channels, opus_app,
&error);
state->in_dtx_mode = 0;
if (error == OPUS_OK && state->encoder != NULL) {
*inst = state;
return 0;
}
free(state);
}
switch (application) {
case 0:
opus_app = OPUS_APPLICATION_VOIP;
break;
case 1:
opus_app = OPUS_APPLICATION_AUDIO;
break;
default:
return -1;
}
return -1;
OpusEncInst* state = calloc(1, sizeof(OpusEncInst));
assert(state);
// Allocate zero counters.
state->zero_counts = calloc(channels, sizeof(size_t));
assert(state->zero_counts);
int error;
state->encoder = opus_encoder_create(48000, channels, opus_app,
&error);
if (error != OPUS_OK || !state->encoder) {
WebRtcOpus_EncoderFree(state);
return -1;
}
state->in_dtx_mode = 0;
state->channels = channels;
*inst = state;
return 0;
}
int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
if (inst) {
opus_encoder_destroy(inst->encoder);
free(inst->zero_counts);
free(inst);
return 0;
} else {
@ -84,13 +98,42 @@ int WebRtcOpus_Encode(OpusEncInst* inst,
size_t length_encoded_buffer,
uint8_t* encoded) {
int res;
size_t i;
int c;
int16_t buffer[2 * 48 * kWebRtcOpusMaxEncodeFrameSizeMs];
if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
return -1;
}
const int channels = inst->channels;
int use_buffer = 0;
// Break long consecutive zeros by forcing a "1" every |kZeroBreakCount|
// samples.
if (inst->in_dtx_mode) {
for (i = 0; i < samples; ++i) {
for (c = 0; c < channels; ++c) {
if (audio_in[i * channels + c] == 0) {
++inst->zero_counts[c];
if (inst->zero_counts[c] == kZeroBreakCount) {
if (!use_buffer) {
memcpy(buffer, audio_in, samples * channels * sizeof(int16_t));
use_buffer = 1;
}
buffer[i * channels + c] = kZeroBreakValue;
inst->zero_counts[c] = 0;
}
} else {
inst->zero_counts[c] = 0;
}
}
}
}
res = opus_encode(inst->encoder,
(const opus_int16*)audio_in,
use_buffer ? buffer : audio_in,
(int)samples,
encoded,
(opus_int32)length_encoded_buffer);