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Adding Opus unit test

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This CL adds a unit test for Opus, as well as new APIs for true stereo decoding (skipping master/slave approach).

BUG=

Review URL: https://webrtc-codereview.appspot.com/1222006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3860 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org
2013-04-17 10:39:41 +00:00
parent 4392d5f9f8
commit db11fab49e
4 changed files with 488 additions and 81 deletions
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259
webrtc/modules/audio_coding/codecs/opus/opus_interface.c
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_coding/codecs/opus/interface/opus_interface.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
#include <stdlib.h>
#include <string.h>
#include "opus.h"
#include "common_audio/signal_processing/resample_by_2_internal.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/common_audio/signal_processing/resample_by_2_internal.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
enum {
/* Maximum supported frame size in WebRTC is 60 ms. */
@ -31,6 +31,9 @@ enum {
/* Sample count is 48 kHz * samples per frame * stereo. */
kWebRtcOpusMaxFrameSize = 48 * kWebRtcOpusMaxDecodeFrameSizeMs * 2,
/* Number of samples in resampler state. */
kWebRtcOpusStateSize = 7,
};
struct WebRtcOpusEncInst {
@ -39,27 +42,34 @@ struct WebRtcOpusEncInst {
int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst, int32_t channels) {
OpusEncInst* state;
state = (OpusEncInst*) calloc(1, sizeof(OpusEncInst));
if (state) {
int error;
// Default to VoIP application for mono, and AUDIO for stereo.
int application = (channels == 1) ?
OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO;
if (inst != NULL) {
state = (OpusEncInst*) calloc(1, sizeof(OpusEncInst));
if (state) {
int error;
/* Default to VoIP application for mono, and AUDIO for stereo. */
int application =
(channels == 1) ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO;
state->encoder = opus_encoder_create(48000, channels, application, &error);
if (error == OPUS_OK || state->encoder != NULL ) {
*inst = state;
return 0;
state->encoder = opus_encoder_create(48000, channels, application,
&error);
if (error == OPUS_OK && state->encoder != NULL) {
*inst = state;
return 0;
}
free(state);
}
free(state);
}
return -1;
}
int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
opus_encoder_destroy(inst->encoder);
free(inst);
return 0;
if (inst) {
opus_encoder_destroy(inst->encoder);
free(inst);
return 0;
} else {
return -1;
}
}
int16_t WebRtcOpus_Encode(OpusEncInst* inst, int16_t* audio_in, int16_t samples,
@ -82,7 +92,11 @@ int16_t WebRtcOpus_Encode(OpusEncInst* inst, int16_t* audio_in, int16_t samples,
}
int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate) {
if (inst) {
return opus_encoder_ctl(inst->encoder, OPUS_SET_BITRATE(rate));
} else {
return -1;
}
}
struct WebRtcOpusDecInst {
@ -98,46 +112,61 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) {
int error_r;
OpusDecInst* state;
// Create Opus decoder memory.
state = (OpusDecInst*) calloc(1, sizeof(OpusDecInst));
if (state == NULL) {
return -1;
}
if (inst != NULL) {
/* Create Opus decoder memory. */
state = (OpusDecInst*) calloc(1, sizeof(OpusDecInst));
if (state == NULL) {
return -1;
}
// Create new memory for left and right channel, always at 48000 Hz.
state->decoder_left = opus_decoder_create(48000, channels, &error_l);
state->decoder_right = opus_decoder_create(48000, channels, &error_r);
if (error_l == OPUS_OK && error_r == OPUS_OK && state->decoder_left != NULL
&& state->decoder_right != NULL) {
// Creation of memory all ok.
state->channels = channels;
*inst = state;
return 0;
}
/* Create new memory for left and right channel, always at 48000 Hz. */
state->decoder_left = opus_decoder_create(48000, channels, &error_l);
state->decoder_right = opus_decoder_create(48000, channels, &error_r);
if (error_l == OPUS_OK && error_r == OPUS_OK && state->decoder_left != NULL
&& state->decoder_right != NULL) {
/* Creation of memory all ok. */
state->channels = channels;
*inst = state;
return 0;
}
// If memory allocation was unsuccessful, free the entire state.
if (state->decoder_left) {
opus_decoder_destroy(state->decoder_left);
/* If memory allocation was unsuccessful, free the entire state. */
if (state->decoder_left) {
opus_decoder_destroy(state->decoder_left);
}
if (state->decoder_right) {
opus_decoder_destroy(state->decoder_right);
}
free(state);
}
if (state->decoder_right) {
opus_decoder_destroy(state->decoder_right);
}
free(state);
state = NULL;
return -1;
}
int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst) {
opus_decoder_destroy(inst->decoder_left);
opus_decoder_destroy(inst->decoder_right);
free(inst);
return 0;
if (inst) {
opus_decoder_destroy(inst->decoder_left);
opus_decoder_destroy(inst->decoder_right);
free(inst);
return 0;
} else {
return -1;
}
}
int WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
return inst->channels;
}
int16_t WebRtcOpus_DecoderInitNew(OpusDecInst* inst) {
int error = opus_decoder_ctl(inst->decoder_left, OPUS_RESET_STATE);
if (error == OPUS_OK) {
memset(inst->state_48_32_left, 0, sizeof(inst->state_48_32_left));
memset(inst->state_48_32_right, 0, sizeof(inst->state_48_32_right));
return 0;
}
return -1;
}
int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) {
int error = opus_decoder_ctl(inst->decoder_left, OPUS_RESET_STATE);
if (error == OPUS_OK) {
@ -156,7 +185,7 @@ int16_t WebRtcOpus_DecoderInitSlave(OpusDecInst* inst) {
return -1;
}
static int DecodeNative(OpusDecoder* inst, int16_t* encoded,
static int DecodeNative(OpusDecoder* inst, const int16_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
unsigned char* coded = (unsigned char*) encoded;
@ -173,16 +202,113 @@ static int DecodeNative(OpusDecoder* inst, int16_t* encoded,
return -1;
}
int16_t WebRtcOpus_Decode(OpusDecInst* inst, int16_t* encoded,
/* Resample from 48 to 32 kHz. Length of state is assumed to be
* kWebRtcOpusStateSize (7).
*/
static int WebRtcOpus_Resample48to32(const int16_t* samples_in, int length,
int16_t* state, int16_t* samples_out) {
int i;
int blocks;
int16_t output_samples;
int32_t buffer32[kWebRtcOpusMaxFrameSize + kWebRtcOpusStateSize];
/* Resample from 48 kHz to 32 kHz. */
for (i = 0; i < kWebRtcOpusStateSize; i++) {
buffer32[i] = state[i];
state[i] = samples_in[length - kWebRtcOpusStateSize + i];
}
for (i = 0; i < length; i++) {
buffer32[kWebRtcOpusStateSize + i] = samples_in[i];
}
/* Resampling 3 samples to 2. Function divides the input in |blocks| number
* of 3-sample groups, and output is |blocks| number of 2-sample groups.
* When this is removed, the compensation in WebRtcOpus_DurationEst should be
* removed too. */
blocks = length / 3;
WebRtcSpl_Resample48khzTo32khz(buffer32, buffer32, blocks);
output_samples = (int16_t) (blocks * 2);
WebRtcSpl_VectorBitShiftW32ToW16(samples_out, output_samples, buffer32, 15);
return output_samples;
}
int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
/* Enough for 120 ms (the largest Opus packet size) of mono audio at 48 kHz
* and resampler overlap. This will need to be enlarged for stereo decoding.
*/
int16_t buffer16_left[kWebRtcOpusMaxFrameSize];
int16_t buffer16_right[kWebRtcOpusMaxFrameSize];
int16_t buffer_out[kWebRtcOpusMaxFrameSize];
int16_t* coded = (int16_t*) encoded;
int decoded_samples;
int resampled_samples;
int i;
/* If mono case, just do a regular call to the decoder.
* If stereo, we need to de-interleave the stereo output in to blocks with
* left and right channel. Each block is resampled to 32 kHz, and then
* interleaved again. */
/* Decode to a temporary buffer. */
decoded_samples = DecodeNative(inst->decoder_left, coded, encoded_bytes,
buffer16_left, audio_type);
if (decoded_samples < 0) {
return -1;
}
/* De-interleave if stereo. */
if (inst->channels == 2) {
/* The parameter |decoded_samples| holds the number of samples pairs, in
* case of stereo. Number of samples in |buffer16| equals |decoded_samples|
* times 2. */
for (i = 0; i < decoded_samples; i++) {
/* Take every second sample, starting at the first sample. */
buffer16_left[i] = buffer16_left[i * 2];
buffer16_right[i] = buffer16_left[i * 2 + 1];
}
/* Resample from 48 kHz to 32 kHz for left channel. */
resampled_samples = WebRtcOpus_Resample48to32(buffer16_left,
decoded_samples,
inst->state_48_32_left,
buffer_out);
/* Add samples interleaved to output vector. */
for (i = 0; i < resampled_samples; i++) {
decoded[i * 2] = buffer_out[i];
}
/* Resample from 48 kHz to 32 kHz for right channel. */
resampled_samples = WebRtcOpus_Resample48to32(buffer16_right,
decoded_samples,
inst->state_48_32_right,
buffer_out);
/* Add samples interleaved to output vector. */
for (i = 0; i < decoded_samples; i++) {
decoded[i * 2 + 1] = buffer_out[i];
}
} else {
/* Resample from 48 kHz to 32 kHz for left channel. */
resampled_samples = WebRtcOpus_Resample48to32(buffer16_left,
decoded_samples,
inst->state_48_32_left,
decoded);
}
return resampled_samples;
}
int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
/* Enough for 120 ms (the largest Opus packet size) of mono audio at 48 kHz
* and resampler overlap. This will need to be enlarged for stereo decoding.
*/
int16_t buffer16[kWebRtcOpusMaxFrameSize];
int32_t buffer32[kWebRtcOpusMaxFrameSize + 7];
int decoded_samples;
int blocks;
int16_t output_samples;
int i;
@ -208,36 +334,22 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, int16_t* encoded,
buffer16[i] = buffer16[i * 2];
}
}
/* Resample from 48 kHz to 32 kHz. */
for (i = 0; i < 7; i++) {
buffer32[i] = inst->state_48_32_left[i];
inst->state_48_32_left[i] = buffer16[decoded_samples - 7 + i];
}
for (i = 0; i < decoded_samples; i++) {
buffer32[7 + i] = buffer16[i];
}
/* Resampling 3 samples to 2. Function divides the input in |blocks| number
* of 3-sample groups, and output is |blocks| number of 2-sample groups.
* When this is removed, the compensation in WebRtcOpus_DurationEst should be
* removed too. */
blocks = decoded_samples / 3;
WebRtcSpl_Resample48khzTo32khz(buffer32, buffer32, blocks);
output_samples = (int16_t) (blocks * 2);
WebRtcSpl_VectorBitShiftW32ToW16(decoded, output_samples, buffer32, 15);
output_samples = WebRtcOpus_Resample48to32(buffer16, decoded_samples,
inst->state_48_32_left, decoded);
return output_samples;
}
int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, int16_t* encoded,
int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const int16_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
/* Enough for 120 ms (the largest Opus packet size) of mono audio at 48 kHz
* and resampler overlap. This will need to be enlarged for stereo decoding.
*/
int16_t buffer16[kWebRtcOpusMaxFrameSize];
int32_t buffer32[kWebRtcOpusMaxFrameSize + 7];
int decoded_samples;
int blocks;
int16_t output_samples;
int i;
@ -261,19 +373,8 @@ int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, int16_t* encoded,
return -1;
}
/* Resample from 48 kHz to 32 kHz. */
for (i = 0; i < 7; i++) {
buffer32[i] = inst->state_48_32_right[i];
inst->state_48_32_right[i] = buffer16[decoded_samples - 7 + i];
}
for (i = 0; i < decoded_samples; i++) {
buffer32[7 + i] = buffer16[i];
}
/* Resampling 3 samples to 2. Function divides the input in |blocks| number
* of 3-sample groups, and output is |blocks| number of 2-sample groups. */
blocks = decoded_samples / 3;
WebRtcSpl_Resample48khzTo32khz(buffer32, buffer32, blocks);
output_samples = (int16_t) (blocks * 2);
WebRtcSpl_VectorBitShiftW32ToW16(decoded, output_samples, buffer32, 15);
output_samples = WebRtcOpus_Resample48to32(buffer16, decoded_samples,
inst->state_48_32_right, decoded);
return output_samples;
}