generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Adding call to Opus PLC

Browse Source 
NetEq will call the PLC function in Opus only to set the decoder state. The actual PLC data will not be used.

BUG=https://code.google.com/p/webrtc/issues/detail?id=1181
R=tterribe@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1727004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4504 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org
2013-08-08 11:01:07 +00:00
parent d177c10e2d
commit bd21fb5f8d
6 changed files with 449 additions and 92 deletions
Download Patch File Download Diff File Expand all files Collapse all files

9
webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
View File

@@ -115,8 +115,11 @@ int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const int16_t* encoded,
int16_t* audio_type);
/****************************************************************************
* WebRtcOpus_DecodePlc(...)
* TODO(tlegrand): Remove master and slave functions when NetEq4 is in place.
* WebRtcOpus_DecodePlcMaster(...)
* WebRtcOpus_DecodePlcSlave(...)
*
* This function precesses PLC for opus frame(s).
* This function processes PLC for opus frame(s).
* Input:
* - inst : Decoder context
* - number_of_lost_frames : Number of PLC frames to produce
@@ -129,6 +132,10 @@ int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const int16_t* encoded,
*/
int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames);
int16_t WebRtcOpus_DecodePlcMaster(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames);
int16_t WebRtcOpus_DecodePlcSlave(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames);
/****************************************************************************
* WebRtcOpus_DurationEst(...)

273
webrtc/modules/audio_coding/codecs/opus/opus_interface.c
View File

@@ -35,8 +35,15 @@ enum {
* milliseconds * maximum number of channels. */
kWebRtcOpusMaxFrameSize = kWebRtcOpusMaxFrameSizePerChannel * 2,
/* Maximum sample count per channel for output resampled to 32 kHz,
* 32 kHz * maximum frame size in milliseconds. */
kWebRtcOpusMaxFrameSizePerChannel32kHz = 32 * kWebRtcOpusMaxDecodeFrameSizeMs,
/* Number of samples in resampler state. */
kWebRtcOpusStateSize = 7,
/* Default frame size, 20 ms @ 48 kHz, in samples (for one channel). */
kWebRtcOpusDefaultFrameSize = 960,
};
struct WebRtcOpusEncInst {
@@ -50,8 +57,8 @@ int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst, int32_t channels) {
if (state) {
int error;
/* Default to VoIP application for mono, and AUDIO for stereo. */
int application =
(channels == 1) ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO;
int application = (channels == 1) ? OPUS_APPLICATION_VOIP :
OPUS_APPLICATION_AUDIO;
state->encoder = opus_encoder_create(48000, channels, application,
&error);
@@ -107,6 +114,7 @@ struct WebRtcOpusDecInst {
int16_t state_48_32_right[8];
OpusDecoder* decoder_left;
OpusDecoder* decoder_right;
int prev_decoded_samples;
int channels;
};
@@ -129,6 +137,7 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) {
&& state->decoder_right != NULL) {
/* Creation of memory all ok. */
state->channels = channels;
state->prev_decoded_samples = kWebRtcOpusDefaultFrameSize;
*inst = state;
return 0;
}
@@ -188,14 +197,17 @@ int16_t WebRtcOpus_DecoderInitSlave(OpusDecInst* inst) {
return -1;
}
/* |frame_size| is set to maximum Opus frame size in the normal case, and
* is set to the number of samples needed for PLC in case of losses.
* It is up to the caller to make sure the value is correct. */
static int DecodeNative(OpusDecoder* inst, const int16_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int16_t encoded_bytes, int frame_size,
int16_t* decoded, int16_t* audio_type) {
unsigned char* coded = (unsigned char*) encoded;
opus_int16* audio = (opus_int16*) decoded;
int res = opus_decode(inst, coded, encoded_bytes, audio,
kWebRtcOpusMaxFrameSizePerChannel, 0);
int res = opus_decode(inst, coded, encoded_bytes, audio, frame_size, 0);
/* TODO(tlegrand): set to DTX for zero-length packets? */
*audio_type = 0;
@@ -213,7 +225,7 @@ static int WebRtcOpus_Resample48to32(const int16_t* samples_in, int length,
int i;
int blocks;
int16_t output_samples;
int32_t buffer32[kWebRtcOpusMaxFrameSize + kWebRtcOpusStateSize];
int32_t buffer32[kWebRtcOpusMaxFrameSizePerChannel + kWebRtcOpusStateSize];
/* Resample from 48 kHz to 32 kHz. */
for (i = 0; i < kWebRtcOpusStateSize; i++) {
@@ -235,75 +247,86 @@ static int WebRtcOpus_Resample48to32(const int16_t* samples_in, int length,
return output_samples;
}
static int WebRtcOpus_DeInterleaveResample(OpusDecInst* inst, int16_t* input,
int sample_pairs, int16_t* output) {
int i;
int16_t buffer_left[kWebRtcOpusMaxFrameSizePerChannel];
int16_t buffer_right[kWebRtcOpusMaxFrameSizePerChannel];
int16_t buffer_out[kWebRtcOpusMaxFrameSizePerChannel32kHz];
int resampled_samples;
/* De-interleave the signal in left and right channel. */
for (i = 0; i < sample_pairs; i++) {
/* Take every second sample, starting at the first sample. */
buffer_left[i] = input[i * 2];
buffer_right[i] = input[i * 2 + 1];
}
/* Resample from 48 kHz to 32 kHz for left channel. */
resampled_samples = WebRtcOpus_Resample48to32(
buffer_left, sample_pairs, inst->state_48_32_left, buffer_out);
/* Add samples interleaved to output vector. */
for (i = 0; i < resampled_samples; i++) {
output[i * 2] = buffer_out[i];
}
/* Resample from 48 kHz to 32 kHz for right channel. */
resampled_samples = WebRtcOpus_Resample48to32(
buffer_right, sample_pairs, inst->state_48_32_right, buffer_out);
/* Add samples interleaved to output vector. */
for (i = 0; i < resampled_samples; i++) {
output[i * 2 + 1] = buffer_out[i];
}
return resampled_samples;
}
int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
/* |buffer16_left| and |buffer_out| are big enough for 120 ms (the largest
* Opus packet size) of stereo audio at 48 kHz, while |buffer16_right| only
* need to be big enough for maximum size of one of the channels. */
int16_t buffer16_left[kWebRtcOpusMaxFrameSize];
int16_t buffer16_right[kWebRtcOpusMaxFrameSizePerChannel];
int16_t buffer_out[kWebRtcOpusMaxFrameSize];
int16_t* coded = (int16_t*) encoded;
/* |buffer| is big enough for 120 ms (the largest Opus packet size) of stereo
* audio at 48 kHz. */
int16_t buffer[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
* If stereo, we need to de-interleave the stereo output into blocks with
* left and right channel. Each block is resampled to 32 kHz, and then
* interleaved again. */
/* Decode to temporarily to |buffer16_left|. */
/* Decode to a temporary buffer. */
decoded_samples = DecodeNative(inst->decoder_left, coded, encoded_bytes,
buffer16_left, audio_type);
kWebRtcOpusMaxFrameSizePerChannel,
buffer, 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_left| 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];
}
/* De-interleave and resample. */
resampled_samples = WebRtcOpus_DeInterleaveResample(inst,
buffer,
decoded_samples,
decoded);
} else {
/* Resample from 48 kHz to 32 kHz for left channel. */
resampled_samples = WebRtcOpus_Resample48to32(buffer16_left,
/* Resample from 48 kHz to 32 kHz. Filter state memory for left channel is
* used for mono signals. */
resampled_samples = WebRtcOpus_Resample48to32(buffer,
decoded_samples,
inst->state_48_32_left,
decoded);
}
/* Update decoded sample memory, to be used by the PLC in case of losses. */
inst->prev_decoded_samples = decoded_samples;
return resampled_samples;
}
int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
@@ -322,7 +345,8 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded,
/* Decode to a temporary buffer. */
decoded_samples = DecodeNative(inst->decoder_left, encoded, encoded_bytes,
buffer16, audio_type);
kWebRtcOpusMaxFrameSizePerChannel, buffer16,
audio_type);
if (decoded_samples < 0) {
return -1;
}
@@ -341,6 +365,9 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded,
output_samples = WebRtcOpus_Resample48to32(buffer16, decoded_samples,
inst->state_48_32_left, decoded);
/* Update decoded sample memory, to be used by the PLC in case of losses. */
inst->prev_decoded_samples = decoded_samples;
return output_samples;
}
@@ -356,7 +383,8 @@ int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const int16_t* encoded,
/* Decode to a temporary buffer. */
decoded_samples = DecodeNative(inst->decoder_right, encoded, encoded_bytes,
buffer16, audio_type);
kWebRtcOpusMaxFrameSizePerChannel, buffer16,
audio_type);
if (decoded_samples < 0) {
return -1;
}
@@ -382,16 +410,141 @@ int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const int16_t* encoded,
int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames) {
/* TODO(tlegrand): We can pass NULL to opus_decode to activate packet
* loss concealment, but I don't know how many samples
* number_of_lost_frames corresponds to. */
return -1;
int16_t buffer[kWebRtcOpusMaxFrameSize];
int16_t audio_type = 0;
int decoded_samples;
int resampled_samples;
int plc_samples;
/* If mono case, just do a regular call to the plc function, before
* resampling.
* If stereo, we need to de-interleave the stereo output into blocks with
* left and right channel. Each block is resampled to 32 kHz, and then
* interleaved again. */
/* Decode to a temporary buffer. The number of samples we ask for is
* |number_of_lost_frames| times |prev_decoded_samples_|. Limit the number
* of samples to maximum |kWebRtcOpusMaxFrameSizePerChannel|. */
plc_samples = number_of_lost_frames * inst->prev_decoded_samples;
plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ?
plc_samples : kWebRtcOpusMaxFrameSizePerChannel;
decoded_samples = DecodeNative(inst->decoder_left, NULL, 0, plc_samples,
buffer, &audio_type);
if (decoded_samples < 0) {
return -1;
}
if (inst->channels == 2) {
/* De-interleave and resample. */
resampled_samples = WebRtcOpus_DeInterleaveResample(inst,
buffer,
decoded_samples,
decoded);
} else {
/* Resample from 48 kHz to 32 kHz. Filter state memory for left channel is
* used for mono signals. */
resampled_samples = WebRtcOpus_Resample48to32(buffer,
decoded_samples,
inst->state_48_32_left,
decoded);
}
return resampled_samples;
}
int16_t WebRtcOpus_DecodePlcMaster(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames) {
int16_t buffer[kWebRtcOpusMaxFrameSize];
int decoded_samples;
int resampled_samples;
int16_t audio_type = 0;
int plc_samples;
int i;
/* If mono case, just do a regular call to the decoder.
* If stereo, call to WebRtcOpus_DecodePlcMaster() gives left channel as
* output, and calls to WebRtcOpus_DecodePlcSlave() give right channel as
* output. This is to make stereo work with the current setup of NetEQ, which
* requires two calls to the decoder to produce stereo. */
/* Decode to a temporary buffer. The number of samples we ask for is
* |number_of_lost_frames| times |prev_decoded_samples_|. Limit the number
* of samples to maximum |kWebRtcOpusMaxFrameSizePerChannel|. */
plc_samples = number_of_lost_frames * inst->prev_decoded_samples;
plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ?
plc_samples : kWebRtcOpusMaxFrameSizePerChannel;
decoded_samples = DecodeNative(inst->decoder_left, NULL, 0, plc_samples,
buffer, &audio_type);
if (decoded_samples < 0) {
return -1;
}
if (inst->channels == 2) {
/* The parameter |decoded_samples| holds the number of sample pairs, in
* case of stereo. The original number of samples in |buffer| equals
* |decoded_samples| times 2. */
for (i = 0; i < decoded_samples; i++) {
/* Take every second sample, starting at the first sample. This gives
* the left channel. */
buffer[i] = buffer[i * 2];
}
}
/* Resample from 48 kHz to 32 kHz for left channel. */
resampled_samples = WebRtcOpus_Resample48to32(buffer,
decoded_samples,
inst->state_48_32_left,
decoded);
return resampled_samples;
}
int16_t WebRtcOpus_DecodePlcSlave(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames) {
int16_t buffer[kWebRtcOpusMaxFrameSize];
int decoded_samples;
int resampled_samples;
int16_t audio_type = 0;
int plc_samples;
int i;
/* Calls to WebRtcOpus_DecodePlcSlave() give right channel as output.
* The function should never be called in the mono case. */
if (inst->channels != 2) {
return -1;
}
/* Decode to a temporary buffer. The number of samples we ask for is
* |number_of_lost_frames| times |prev_decoded_samples_|. Limit the number
* of samples to maximum |kWebRtcOpusMaxFrameSizePerChannel|. */
plc_samples = number_of_lost_frames * inst->prev_decoded_samples;
plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel)
? plc_samples : kWebRtcOpusMaxFrameSizePerChannel;
decoded_samples = DecodeNative(inst->decoder_right, NULL, 0, plc_samples,
buffer, &audio_type);
if (decoded_samples < 0) {
return -1;
}
/* The parameter |decoded_samples| holds the number of sample pairs,
* The original number of samples in |buffer| equals |decoded_samples|
* times 2. */
for (i = 0; i < decoded_samples; i++) {
/* Take every second sample, starting at the second sample. This gives
* the right channel. */
buffer[i] = buffer[i * 2 + 1];
}
/* Resample from 48 kHz to 32 kHz for left channel. */
resampled_samples = WebRtcOpus_Resample48to32(buffer,
decoded_samples,
inst->state_48_32_right,
decoded);
return resampled_samples;
}
int WebRtcOpus_DurationEst(OpusDecInst* inst,
const uint8_t* payload,
int payload_length_bytes)
{
int payload_length_bytes) {
int frames, samples;
frames = opus_packet_get_nb_frames(payload, payload_length_bytes);
if (frames < 0) {

124
webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
View File

@@ -117,8 +117,8 @@ TEST_F(OpusTest, OpusEncodeDecodeMono) {
int16_t output_data_decode_new[kOpusNumberOfSamples];
int16_t output_data_decode[kOpusNumberOfSamples];
int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_, 960,
kMaxBytes, bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_, 960,
kMaxBytes, bitstream_);
EXPECT_EQ(640, WebRtcOpus_DecodeNew(opus_mono_decoder_new_, bitstream_,
encoded_bytes, output_data_decode_new,
&audio_type));
@@ -158,8 +158,8 @@ TEST_F(OpusTest, OpusEncodeDecodeStereo) {
int16_t output_data_decode[kOpusNumberOfSamples];
int16_t output_data_decode_slave[kOpusNumberOfSamples];
int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 960,
kMaxBytes, bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 960,
kMaxBytes, bitstream_);
EXPECT_EQ(640, WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_,
encoded_bytes, output_data_decode_new,
&audio_type));
@@ -217,8 +217,8 @@ TEST_F(OpusTest, OpusDecodeInit) {
int16_t output_data_decode[kOpusNumberOfSamples];
int16_t output_data_decode_slave[kOpusNumberOfSamples];
int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 960,
kMaxBytes, bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 960,
kMaxBytes, bitstream_);
EXPECT_EQ(640, WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_,
encoded_bytes, output_data_decode_new,
&audio_type));
@@ -265,10 +265,108 @@ TEST_F(OpusTest, OpusDecodeInit) {
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_stereo_decoder_new_));
}
// PLC not implemented.
TEST_F(OpusTest, OpusDecodePlc) {
// PLC in mono mode.
TEST_F(OpusTest, OpusDecodePlcMono) {
// Create encoder memory.
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_mono_decoder_new_, 1));
// Set bitrate.
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, 32000));
// Check number of channels for decoder.
EXPECT_EQ(1, WebRtcOpus_DecoderChannels(opus_mono_decoder_));
EXPECT_EQ(1, WebRtcOpus_DecoderChannels(opus_mono_decoder_new_));
// Encode & decode.
int16_t encoded_bytes;
int16_t audio_type;
int16_t output_data_decode_new[kOpusNumberOfSamples];
int16_t output_data_decode[kOpusNumberOfSamples];
int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_, 960,
kMaxBytes, bitstream_);
EXPECT_EQ(640, WebRtcOpus_DecodeNew(opus_mono_decoder_new_, bitstream_,
encoded_bytes, output_data_decode_new,
&audio_type));
EXPECT_EQ(640, WebRtcOpus_Decode(opus_mono_decoder_, coded,
encoded_bytes, output_data_decode,
&audio_type));
// Call decoder PLC for both versions of the decoder.
int16_t plc_buffer[kOpusNumberOfSamples];
EXPECT_EQ(-1, WebRtcOpus_DecodePlc(opus_stereo_decoder_, plc_buffer, 1));
int16_t plc_buffer_new[kOpusNumberOfSamples];
EXPECT_EQ(640, WebRtcOpus_DecodePlcMaster(opus_mono_decoder_, plc_buffer, 1));
EXPECT_EQ(640, WebRtcOpus_DecodePlc(opus_mono_decoder_new_,
plc_buffer_new, 1));
// Data in |plc_buffer| should be the same as in |plc_buffer_new|.
for (int i = 0; i < 640; i++) {
EXPECT_EQ(plc_buffer[i], plc_buffer_new[i]);
}
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_mono_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_mono_decoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_mono_decoder_new_));
}
// PLC in stereo mode.
TEST_F(OpusTest, OpusDecodePlcStereo) {
// Create encoder memory.
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_stereo_decoder_new_, 2));
// Set bitrate.
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, 64000));
// Check number of channels for decoder.
EXPECT_EQ(2, WebRtcOpus_DecoderChannels(opus_stereo_decoder_));
EXPECT_EQ(2, WebRtcOpus_DecoderChannels(opus_stereo_decoder_new_));
// Encode & decode.
int16_t encoded_bytes;
int16_t audio_type;
int16_t output_data_decode_new[kOpusNumberOfSamples];
int16_t output_data_decode[kOpusNumberOfSamples];
int16_t output_data_decode_slave[kOpusNumberOfSamples];
int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 960,
kMaxBytes, bitstream_);
EXPECT_EQ(640, WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_,
encoded_bytes, output_data_decode_new,
&audio_type));
EXPECT_EQ(640, WebRtcOpus_Decode(opus_stereo_decoder_, coded,
encoded_bytes, output_data_decode,
&audio_type));
EXPECT_EQ(640, WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded,
encoded_bytes,
output_data_decode_slave,
&audio_type));
// Call decoder PLC for both versions of the decoder.
int16_t plc_buffer_left[kOpusNumberOfSamples];
int16_t plc_buffer_right[kOpusNumberOfSamples];
int16_t plc_buffer_new[kOpusNumberOfSamples];
EXPECT_EQ(640, WebRtcOpus_DecodePlcMaster(opus_stereo_decoder_,
plc_buffer_left, 1));
EXPECT_EQ(640, WebRtcOpus_DecodePlcSlave(opus_stereo_decoder_,
plc_buffer_right, 1));
EXPECT_EQ(640, WebRtcOpus_DecodePlc(opus_stereo_decoder_new_, plc_buffer_new,
1));
// Data in |plc_buffer_left| and |plc_buffer_right|should be the same as the
// interleaved samples in |plc_buffer_new|.
for (int i = 0, j = 0; i < 640; i++) {
EXPECT_EQ(plc_buffer_left[i], plc_buffer_new[j++]);
EXPECT_EQ(plc_buffer_right[i], plc_buffer_new[j++]);
}
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_stereo_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_stereo_decoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_stereo_decoder_new_));
}
// Duration estimation.
@@ -281,14 +379,14 @@ TEST_F(OpusTest, OpusDurationEstimation) {
int16_t encoded_bytes;
// 10 ms.
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 480,
kMaxBytes, bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 480,
kMaxBytes, bitstream_);
EXPECT_EQ(320, WebRtcOpus_DurationEst(opus_stereo_decoder_, bitstream_,
encoded_bytes));
// 20 ms
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 960,
kMaxBytes, bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 960,
kMaxBytes, bitstream_);
EXPECT_EQ(640, WebRtcOpus_DurationEst(opus_stereo_decoder_, bitstream_,
encoded_bytes));