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 2ac53736650907ca4d54cfb1e1ccc42646f1116a..f1dfe214b632df65daf1952e898e21b5737432c0 100644 |
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c |
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c |
@@ -29,48 +29,63 @@ 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, |
}; |
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; |
- } |
- } |
+ if (inst == NULL) |
kwiberg-webrtc
2015/10/29 16:35:21
if (!inst)
minyue-webrtc
2015/10/30 14:06:46
Done.
|
+ 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); |
+ OpusEncInst* state = (OpusEncInst*)calloc(1, sizeof(OpusEncInst)); |
kwiberg-webrtc
2015/10/29 16:35:21
You don't need to cast. C allows implicit conversi
minyue-webrtc
2015/10/30 14:06:46
Done.
|
+ if (!state) |
+ return -1; |
+ |
+ // Allocate zero counters. |
+ state->zero_counts = (size_t*)calloc(channels, sizeof(size_t)); |
kwiberg-webrtc
2015/10/29 16:35:21
No need to cast.
minyue-webrtc
2015/10/30 14:06:46
Done.
|
+ if (!state->zero_counts) { |
+ free(state); |
+ return -1; |
kwiberg-webrtc
2015/10/29 16:35:21
Don't try to handle malloc failures. Just assert i
minyue-webrtc
2015/10/30 14:06:46
Done.
|
+ } |
+ |
+ int opus_app; |
+ switch (application) { |
+ case 0: { |
+ opus_app = OPUS_APPLICATION_VOIP; |
+ break; |
+ } |
+ case 1: { |
+ opus_app = OPUS_APPLICATION_AUDIO; |
+ break; |
+ } |
+ default: { |
+ WebRtcOpus_EncoderFree(state); |
minyue-webrtc
2015/10/30 14:06:46
I found switch() {} can be placed before state get
|
+ return -1; |
} |
} |
- return -1; |
+ |
+ int error; |
+ state->encoder = opus_encoder_create(48000, channels, opus_app, |
+ &error); |
+ if (error != OPUS_OK || state->encoder == NULL) { |
kwiberg-webrtc
2015/10/29 16:35:21
!state->encoder
minyue-webrtc
2015/10/30 14:06:46
Done.
|
+ 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 +99,35 @@ int WebRtcOpus_Encode(OpusEncInst* inst, |
size_t length_encoded_buffer, |
uint8_t* encoded) { |
int res; |
+ int16_t buffer[2 * 48 * kWebRtcOpusMaxEncodeFrameSizeMs]; |
if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) { |
return -1; |
} |
+ const int channels = inst->channels; |
+ int16_t* pointer = buffer; |
+ // Break long consecutive zeros by forcing a "1" every |kZeroBreakCount| |
+ // samples. |
+ memcpy(pointer, audio_in, samples * channels * sizeof(int16_t)); |
+ if (inst->in_dtx_mode) { |
+ for (size_t i = 0; i < samples; ++i) { |
+ for (int c = 0; c < channels; ++c, ++pointer) { |
+ if (*pointer == 0) { |
+ ++inst->zero_counts[c]; |
+ if (inst->zero_counts[c] == kZeroBreakCount) { |
+ *pointer = 1; |
+ inst->zero_counts[c] = 0; |
+ } |
+ } else { |
+ inst->zero_counts[c] = 0; |
+ } |
+ } |
+ } |
+ } |
kwiberg-webrtc
2015/10/29 16:35:21
It would've been easier to read this loop if you'd
minyue-webrtc
2015/10/30 14:06:46
Yes, I agree that "input[i * channels + c]" and I
kwiberg-webrtc
2015/11/01 02:01:55
Yes. I tried these two:
void f1(int a, int b) {
|
+ |
res = opus_encode(inst->encoder, |
- (const opus_int16*)audio_in, |
+ buffer, |
(int)samples, |
encoded, |
(opus_int32)length_encoded_buffer); |