| 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);
|
|
|