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 1a632422c5e150facf1ceb4d072b2df70ba3fd4c..9eee89f1327fc0e841d2448ff288ac4deae19aee 100644 |
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c |
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c |
@@ -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); |