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Unified Diff: webrtc/modules/audio_coding/codecs/opus/opus_interface.c

Issue 1428613004: Revert of Prevent Opus DTX from generating intermittent noise during silence (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 1 month ago
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Index: webrtc/modules/audio_coding/codecs/opus/opus_interface.c
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index 9eee89f1327fc0e841d2448ff288ac4deae19aee..1a632422c5e150facf1ceb4d072b2df70ba3fd4c 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
@@ -11,7 +11,6 @@
#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>
@@ -30,61 +29,48 @@
/* 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) {
- int opus_app;
- if (!inst)
- return -1;
-
- switch (application) {
- case 0:
- opus_app = OPUS_APPLICATION_VOIP;
- break;
- case 1:
- opus_app = OPUS_APPLICATION_AUDIO;
- break;
- default:
- 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;
+ 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 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);
+ }
+ }
+ return -1;
}
int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
if (inst) {
opus_encoder_destroy(inst->encoder);
- free(inst->zero_counts);
free(inst);
return 0;
} else {
@@ -98,42 +84,13 @@
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,
- use_buffer ? buffer : audio_in,
+ (const opus_int16*)audio_in,
(int)samples,
encoded,
(opus_int32)length_encoded_buffer);
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