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Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.

Browse Source 
TEST=passed_all_trybots
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6458 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
minyue@webrtc.org 2014-06-17 08:02:05 +00:00
parent 44a317a698
commit 8f8503d947
12 changed files with 249 additions and 433 deletions
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35
webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
View File

@ -32,8 +32,7 @@ struct mode {
};
const int kOpusBlockDurationMs = 20;
const int kOpusInputSamplingKhz = 48;
const int kOpusOutputSamplingKhz = 32;
const int kOpusSamplingKhz = 48;
class OpusFecTest : public TestWithParam<coding_param> {
protected:
@ -47,14 +46,8 @@ class OpusFecTest : public TestWithParam<coding_param> {
virtual void DecodeABlock(bool lost_previous, bool lost_current);
int block_duration_ms_;
int input_sampling_khz_;
int output_sampling_khz_;
// Number of samples-per-channel in a frame.
int input_length_sample_;
// Expected output number of samples-per-channel in a frame.
int output_length_sample_;
int sampling_khz_;
int block_length_sample_;
int channels_;
int bit_rate_;
@ -91,7 +84,7 @@ void OpusFecTest::SetUp() {
// Allocate memory to contain the whole file.
in_data_.reset(new int16_t[loop_length_samples_ +
input_length_sample_ * channels_]);
block_length_sample_ * channels_]);
// Copy the file into the buffer.
ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
@ -104,12 +97,12 @@ void OpusFecTest::SetUp() {
// beginning of the array. Audio frames cross the end of the excerpt always
// appear as a continuum of memory.
memcpy(&in_data_[loop_length_samples_], &in_data_[0],
input_length_sample_ * channels_ * sizeof(int16_t));
block_length_sample_ * channels_ * sizeof(int16_t));
// Maximum number of bytes in output bitstream.
max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t);
max_bytes_ = block_length_sample_ * channels_ * sizeof(int16_t);
out_data_.reset(new int16_t[2 * output_length_sample_ * channels_]);
out_data_.reset(new int16_t[2 * block_length_sample_ * channels_]);
bit_stream_.reset(new uint8_t[max_bytes_]);
// Create encoder memory.
@ -127,10 +120,8 @@ void OpusFecTest::TearDown() {
OpusFecTest::OpusFecTest()
: block_duration_ms_(kOpusBlockDurationMs),
input_sampling_khz_(kOpusInputSamplingKhz),
output_sampling_khz_(kOpusOutputSamplingKhz),
input_length_sample_(block_duration_ms_ * input_sampling_khz_),
output_length_sample_(block_duration_ms_ * output_sampling_khz_),
sampling_khz_(kOpusSamplingKhz),
block_length_sample_(block_duration_ms_ * sampling_khz_),
data_pointer_(0),
max_bytes_(0),
encoded_bytes_(0),
@ -141,7 +132,7 @@ OpusFecTest::OpusFecTest()
void OpusFecTest::EncodeABlock() {
int16_t value = WebRtcOpus_Encode(opus_encoder_,
&in_data_[data_pointer_],
input_length_sample_,
block_length_sample_,
max_bytes_, &bit_stream_[0]);
EXPECT_GT(value, 0);
@ -162,7 +153,7 @@ void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
} else {
value_1 = WebRtcOpus_DecodePlc(opus_decoder_, &out_data_[0], 1);
}
EXPECT_EQ(output_length_sample_, value_1);
EXPECT_EQ(block_length_sample_, value_1);
}
if (!lost_current) {
@ -171,7 +162,7 @@ void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
encoded_bytes_,
&out_data_[value_1 * channels_],
&audio_type);
EXPECT_EQ(output_length_sample_, value_2);
EXPECT_EQ(block_length_sample_, value_2);
}
}
@ -224,7 +215,7 @@ TEST_P(OpusFecTest, RandomPacketLossTest) {
// |data_pointer_| is incremented and wrapped across
// |loop_length_samples_|.
data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) %
data_pointer_ = (data_pointer_ + block_length_sample_ * channels_) %
loop_length_samples_;
}
if (mode_set[i].fec) {

259
webrtc/modules/audio_coding/codecs/opus/opus_interface.c
View File

@ -15,9 +15,6 @@
#include "opus.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. */
kWebRtcOpusMaxEncodeFrameSizeMs = 60,
@ -31,17 +28,6 @@ enum {
* milliseconds. */
kWebRtcOpusMaxFrameSizePerChannel = 48 * kWebRtcOpusMaxDecodeFrameSizeMs,
/* Maximum sample count per frame is 48 kHz * maximum frame size in
* 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,
};
@ -143,8 +129,6 @@ int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) {
}
struct WebRtcOpusDecInst {
int16_t state_48_32_left[8];
int16_t state_48_32_right[8];
OpusDecoder* decoder_left;
OpusDecoder* decoder_right;
int prev_decoded_samples;
@ -205,8 +189,6 @@ int WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
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;
@ -215,7 +197,6 @@ int16_t WebRtcOpus_DecoderInitNew(OpusDecInst* inst) {
int16_t WebRtcOpus_DecoderInit(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));
return 0;
}
return -1;
@ -224,7 +205,6 @@ int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) {
int16_t WebRtcOpus_DecoderInitSlave(OpusDecInst* inst) {
int error = opus_decoder_ctl(inst->decoder_right, OPUS_RESET_STATE);
if (error == OPUS_OK) {
memset(inst->state_48_32_right, 0, sizeof(inst->state_48_32_right));
return 0;
}
return -1;
@ -267,124 +247,29 @@ static int DecodeFec(OpusDecoder* inst, const int16_t* encoded,
return -1;
}
/* 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[kWebRtcOpusMaxFrameSizePerChannel + 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;
}
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) {
/* |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;
/* If mono case, just do a regular call to the decoder.
* 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. */
decoded_samples = DecodeNative(inst->decoder_left, coded, encoded_bytes,
kWebRtcOpusMaxFrameSizePerChannel,
buffer, audio_type);
decoded, 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);
}
/* Update decoded sample memory, to be used by the PLC in case of losses. */
inst->prev_decoded_samples = decoded_samples;
return resampled_samples;
return decoded_samples;
}
int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
/* |buffer16| is big enough for 120 ms (the largestOpus packet size) of
* stereo audio at 48 kHz. */
int16_t buffer16[kWebRtcOpusMaxFrameSize];
int decoded_samples;
int16_t output_samples;
int i;
/* If mono case, just do a regular call to the decoder.
@ -393,120 +278,82 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded,
* 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. */
decoded_samples = DecodeNative(inst->decoder_left, encoded, encoded_bytes,
kWebRtcOpusMaxFrameSizePerChannel, buffer16,
kWebRtcOpusMaxFrameSizePerChannel, decoded,
audio_type);
if (decoded_samples < 0) {
return -1;
}
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|
* case of stereo. Number of samples in |decoded| 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. */
buffer16[i] = buffer16[i * 2];
decoded[i] = decoded[i * 2];
}
}
/* Resample from 48 kHz to 32 kHz. */
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;
return decoded_samples;
}
int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const int16_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
/* |buffer16| is big enough for 120 ms (the largestOpus packet size) of
* stereo audio at 48 kHz. */
int16_t buffer16[kWebRtcOpusMaxFrameSize];
int decoded_samples;
int16_t output_samples;
int i;
/* Decode to a temporary buffer. */
decoded_samples = DecodeNative(inst->decoder_right, encoded, encoded_bytes,
kWebRtcOpusMaxFrameSizePerChannel, buffer16,
kWebRtcOpusMaxFrameSizePerChannel, decoded,
audio_type);
if (decoded_samples < 0) {
return -1;
}
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|
* case of stereo. Number of samples in |decoded| 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. */
buffer16[i] = buffer16[i * 2 + 1];
decoded[i] = decoded[i * 2 + 1];
}
} else {
/* Decode slave should never be called for mono packets. */
return -1;
}
/* Resample from 48 kHz to 32 kHz. */
output_samples = WebRtcOpus_Resample48to32(buffer16, decoded_samples,
inst->state_48_32_right, decoded);
return output_samples;
return decoded_samples;
}
int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames) {
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|. */
/* 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);
decoded, &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;
return decoded_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;
@ -517,42 +364,35 @@ int16_t WebRtcOpus_DecodePlcMaster(OpusDecInst* inst, int16_t* decoded,
* 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|. */
/* 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);
decoded, &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
* case of stereo. The original number of samples in |decoded| 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];
decoded[i] = decoded[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;
return decoded_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;
@ -563,44 +403,35 @@ int16_t WebRtcOpus_DecodePlcSlave(OpusDecInst* inst, int16_t* decoded,
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|. */
/* 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);
decoded, &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|
* The original number of samples in |decoded| 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];
decoded[i] = decoded[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;
return decoded_samples;
}
int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
/* |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 fec_samples;
if (WebRtcOpus_PacketHasFec(encoded, encoded_bytes) != 1) {
@ -609,33 +440,13 @@ int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
fec_samples = opus_packet_get_samples_per_frame(encoded, 48000);
/* Decode to a temporary buffer. */
decoded_samples = DecodeFec(inst->decoder_left, coded, encoded_bytes,
fec_samples, buffer, audio_type);
fec_samples, decoded, audio_type);
if (decoded_samples < 0) {
return -1;
}
/* If mono case, just do a regular call to the decoder.
* 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. */
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;
return decoded_samples;
}
int WebRtcOpus_DurationEst(OpusDecInst* inst,
@ -652,10 +463,6 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst,
/* Invalid payload duration. */
return 0;
}
/* Compensate for the down-sampling from 48 kHz to 32 kHz.
* This should be removed when the resampling in WebRtcOpus_Decode is
* removed. */
samples = samples * 2 / 3;
return samples;
}
@ -671,10 +478,6 @@ int WebRtcOpus_FecDurationEst(const uint8_t* payload,
/* Invalid payload duration. */
return 0;
}
/* Compensate for the down-sampling from 48 kHz to 32 kHz.
* This should be removed when the resampling in WebRtcOpus_Decode is
* removed. */
samples = samples * 2 / 3;
return samples;
}

7
webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
View File

@ -18,8 +18,7 @@ using ::testing::ValuesIn;
namespace webrtc {
static const int kOpusBlockDurationMs = 20;
static const int kOpusInputSamplingKhz = 48;
static const int kOpustOutputSamplingKhz = 32;
static const int kOpusSamplingKhz = 48;
class OpusSpeedTest : public AudioCodecSpeedTest {
protected:
@ -36,8 +35,8 @@ class OpusSpeedTest : public AudioCodecSpeedTest {
OpusSpeedTest::OpusSpeedTest()
: AudioCodecSpeedTest(kOpusBlockDurationMs,
kOpusInputSamplingKhz,
kOpustOutputSamplingKhz),
kOpusSamplingKhz,
kOpusSamplingKhz),
opus_encoder_(NULL),
opus_decoder_(NULL) {
}

241
webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
View File

@ -19,9 +19,13 @@ struct WebRtcOpusDecInst;
namespace webrtc {
// Number of samples in a 60 ms stereo frame, sampled at 48 kHz.
const int kOpusNumberOfSamples = 480 * 6 * 2;
const int kOpusMaxFrameSamples = 48 * 60 * 2;
// Maximum number of bytes in output bitstream.
const size_t kMaxBytes = 1000;
// Number of samples-per-channel in a 20 ms frame, sampled at 48 kHz.
const int kOpus20msFrameSamples = 48 * 20;
// Number of samples-per-channel in a 10 ms frame, sampled at 48 kHz.
const int kOpus10msFrameSamples = 48 * 10;
class OpusTest : public ::testing::Test {
protected:
@ -35,8 +39,8 @@ class OpusTest : public ::testing::Test {
WebRtcOpusDecInst* opus_stereo_decoder_;
WebRtcOpusDecInst* opus_stereo_decoder_new_;
int16_t speech_data_[kOpusNumberOfSamples];
int16_t output_data_[kOpusNumberOfSamples];
int16_t speech_data_[kOpusMaxFrameSamples];
int16_t output_data_[kOpusMaxFrameSamples];
uint8_t bitstream_[kMaxBytes];
};
@ -50,17 +54,14 @@ OpusTest::OpusTest()
}
void OpusTest::SetUp() {
// Read some samples from a speech file, to be used in the encode test.
// In this test we do not care that the sampling frequency of the file is
// really 32000 Hz. We pretend that it is 48000 Hz.
FILE* input_file;
const std::string file_name =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
input_file = fopen(file_name.c_str(), "rb");
ASSERT_TRUE(input_file != NULL);
ASSERT_EQ(kOpusNumberOfSamples,
ASSERT_EQ(kOpusMaxFrameSamples,
static_cast<int32_t>(fread(speech_data_, sizeof(int16_t),
kOpusNumberOfSamples, input_file)));
kOpusMaxFrameSamples, input_file)));
fclose(input_file);
input_file = NULL;
}
@ -114,21 +115,24 @@ TEST_F(OpusTest, OpusEncodeDecodeMono) {
// 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_new[kOpusMaxFrameSamples];
int16_t output_data_decode[kOpusMaxFrameSamples];
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));
encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes,
bitstream_);
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodeNew(opus_mono_decoder_new_, bitstream_,
encoded_bytes, output_data_decode_new,
&audio_type));
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_mono_decoder_, coded,
encoded_bytes, output_data_decode,
&audio_type));
// Data in |output_data_decode_new| should be the same as in
// |output_data_decode|.
for (int i = 0; i < 640; i++) {
for (int i = 0; i < kOpus20msFrameSamples; i++) {
EXPECT_EQ(output_data_decode_new[i], output_data_decode[i]);
}
@ -154,26 +158,30 @@ TEST_F(OpusTest, OpusEncodeDecodeStereo) {
// 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 output_data_decode_new[kOpusMaxFrameSamples];
int16_t output_data_decode[kOpusMaxFrameSamples];
int16_t output_data_decode_slave[kOpusMaxFrameSamples];
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,
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes,
bitstream_);
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_,
encoded_bytes, output_data_decode_new,
&audio_type));
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_stereo_decoder_, coded,
encoded_bytes, output_data_decode,
&audio_type));
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded,
encoded_bytes, output_data_decode_slave,
&audio_type));
EXPECT_EQ(640, WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded,
encoded_bytes, output_data_decode_slave,
&audio_type));
// Data in |output_data_decode_new| should be the same as in
// |output_data_decode| and |output_data_decode_slave| interleaved to a
// stereo signal.
for (int i = 0; i < 640; i++) {
for (int i = 0; i < kOpus20msFrameSamples; i++) {
EXPECT_EQ(output_data_decode_new[i * 2], output_data_decode[i]);
EXPECT_EQ(output_data_decode_new[i * 2 + 1], output_data_decode_slave[i]);
}
@ -234,26 +242,30 @@ TEST_F(OpusTest, OpusDecodeInit) {
// 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 output_data_decode_new[kOpusMaxFrameSamples];
int16_t output_data_decode[kOpusMaxFrameSamples];
int16_t output_data_decode_slave[kOpusMaxFrameSamples];
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,
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes,
bitstream_);
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_,
encoded_bytes, output_data_decode_new,
&audio_type));
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_stereo_decoder_, coded,
encoded_bytes, output_data_decode,
&audio_type));
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded,
encoded_bytes, output_data_decode_slave,
&audio_type));
EXPECT_EQ(640, WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded,
encoded_bytes, output_data_decode_slave,
&audio_type));
// Data in |output_data_decode_new| should be the same as in
// |output_data_decode| and |output_data_decode_slave| interleaved to a
// stereo signal.
for (int i = 0; i < 640; i++) {
for (int i = 0; i < kOpus20msFrameSamples; i++) {
EXPECT_EQ(output_data_decode_new[i * 2], output_data_decode[i]);
EXPECT_EQ(output_data_decode_new[i * 2 + 1], output_data_decode_slave[i]);
}
@ -262,20 +274,23 @@ TEST_F(OpusTest, OpusDecodeInit) {
EXPECT_EQ(0, WebRtcOpus_DecoderInit(opus_stereo_decoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderInitSlave(opus_stereo_decoder_));
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,
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_,
encoded_bytes, output_data_decode_new,
&audio_type));
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_stereo_decoder_, coded,
encoded_bytes, output_data_decode,
&audio_type));
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded,
encoded_bytes, output_data_decode_slave,
&audio_type));
EXPECT_EQ(640, WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded,
encoded_bytes, output_data_decode_slave,
&audio_type));
// Data in |output_data_decode_new| should be the same as in
// |output_data_decode| and |output_data_decode_slave| interleaved to a
// stereo signal.
for (int i = 0; i < 640; i++) {
for (int i = 0; i < kOpus20msFrameSamples; i++) {
EXPECT_EQ(output_data_decode_new[i * 2], output_data_decode[i]);
EXPECT_EQ(output_data_decode_new[i * 2 + 1], output_data_decode_slave[i]);
}
@ -344,27 +359,31 @@ TEST_F(OpusTest, OpusDecodePlcMono) {
// 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_new[kOpusMaxFrameSamples];
int16_t output_data_decode[kOpusMaxFrameSamples];
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));
encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes,
bitstream_);
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodeNew(opus_mono_decoder_new_, bitstream_,
encoded_bytes, output_data_decode_new,
&audio_type));
EXPECT_EQ(kOpus20msFrameSamples,
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];
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));
int16_t plc_buffer[kOpusMaxFrameSamples];
int16_t plc_buffer_new[kOpusMaxFrameSamples];
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodePlcMaster(opus_mono_decoder_, plc_buffer, 1));
EXPECT_EQ(kOpus20msFrameSamples,
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++) {
for (int i = 0; i < kOpus20msFrameSamples; i++) {
EXPECT_EQ(plc_buffer[i], plc_buffer_new[i]);
}
@ -391,36 +410,42 @@ TEST_F(OpusTest, OpusDecodePlcStereo) {
// 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 output_data_decode_new[kOpusMaxFrameSamples];
int16_t output_data_decode[kOpusMaxFrameSamples];
int16_t output_data_decode_slave[kOpusMaxFrameSamples];
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,
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes,
bitstream_);
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_,
encoded_bytes, output_data_decode_new,
&audio_type));
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_stereo_decoder_, coded,
encoded_bytes, output_data_decode,
&audio_type));
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded,
encoded_bytes,
output_data_decode_slave,
&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));
int16_t plc_buffer_left[kOpusMaxFrameSamples];
int16_t plc_buffer_right[kOpusMaxFrameSamples];
int16_t plc_buffer_new[kOpusMaxFrameSamples];
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodePlcMaster(opus_stereo_decoder_,
plc_buffer_left, 1));
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodePlcSlave(opus_stereo_decoder_,
plc_buffer_right, 1));
EXPECT_EQ(kOpus20msFrameSamples,
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++) {
for (int i = 0, j = 0; i < kOpus20msFrameSamples; i++) {
EXPECT_EQ(plc_buffer_left[i], plc_buffer_new[j++]);
EXPECT_EQ(plc_buffer_right[i], plc_buffer_new[j++]);
}
@ -437,21 +462,23 @@ TEST_F(OpusTest, OpusDurationEstimation) {
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2));
// Encode with different packet sizes (input 48 kHz, output in 32 kHz).
int16_t encoded_bytes;
// 10 ms.
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 480,
kMaxBytes, bitstream_);
EXPECT_EQ(320, WebRtcOpus_DurationEst(opus_stereo_decoder_, bitstream_,
encoded_bytes));
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
kOpus10msFrameSamples, kMaxBytes,
bitstream_);
EXPECT_EQ(kOpus10msFrameSamples,
WebRtcOpus_DurationEst(opus_stereo_decoder_, bitstream_,
encoded_bytes));
// 20 ms
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, 960,
kMaxBytes, bitstream_);
EXPECT_EQ(640, WebRtcOpus_DurationEst(opus_stereo_decoder_, bitstream_,
encoded_bytes));
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes,
bitstream_);
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DurationEst(opus_stereo_decoder_, bitstream_,
encoded_bytes));
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_stereo_encoder_));

10
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
View File

@ -1616,14 +1616,8 @@ int AudioCodingModuleImpl::ReceiveFrequency() const {
int codec_id = receiver_.last_audio_codec_id();
int sample_rate_hz;
if (codec_id < 0)
sample_rate_hz = receiver_.current_sample_rate_hz();
else
sample_rate_hz = ACMCodecDB::database_[codec_id].plfreq;
// TODO(tlegrand): Remove this option when we have full 48 kHz support.
return (sample_rate_hz > 32000) ? 32000 : sample_rate_hz;
return codec_id < 0 ? receiver_.current_sample_rate_hz() :
ACMCodecDB::database_[codec_id].plfreq;
}
// Get current playout frequency.

18
webrtc/modules/audio_coding/main/test/opus_test.cc
View File

@ -218,6 +218,8 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
int written_samples = 0;
int read_samples = 0;
int decoded_samples = 0;
bool first_packet = true;
uint32_t start_time_stamp = 0;
channel->reset_payload_size();
counter_ = 0;
@ -324,6 +326,10 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
// Send data to the channel. "channel" will handle the loss simulation.
channel->SendData(kAudioFrameSpeech, payload_type_, rtp_timestamp_,
bitstream, bitstream_len_byte, NULL);
if (first_packet) {
first_packet = false;
start_time_stamp = rtp_timestamp_;
}
rtp_timestamp_ += frame_length;
read_samples += frame_length * channels;
}
@ -344,9 +350,11 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
// Write stand-alone speech to file.
out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels);
// Number of channels should be the same for both stand-alone and
// ACM-decoding.
EXPECT_EQ(audio_frame.num_channels_, channels);
if (audio_frame.timestamp_ > start_time_stamp) {
// Number of channels should be the same for both stand-alone and
// ACM-decoding.
EXPECT_EQ(audio_frame.num_channels_, channels);
}
decoded_samples = 0;
}
@ -367,13 +375,13 @@ void OpusTest::OpenOutFile(int test_number) {
file_stream << webrtc::test::OutputPath() << "opustest_out_"
<< test_number << ".pcm";
file_name = file_stream.str();
out_file_.Open(file_name, 32000, "wb");
out_file_.Open(file_name, 48000, "wb");
file_stream.str("");
file_name = file_stream.str();
file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
<< test_number << ".pcm";
file_name = file_stream.str();
out_file_standalone_.Open(file_name, 32000, "wb");
out_file_standalone_.Open(file_name, 48000, "wb");
}
} // namespace webrtc

2
webrtc/modules/audio_coding/neteq/audio_decoder.cc
View File

@ -162,7 +162,7 @@ int AudioDecoder::CodecSampleRateHz(NetEqDecoder codec_type) {
#ifdef WEBRTC_CODEC_OPUS
case kDecoderOpus:
case kDecoderOpus_2ch: {
return 32000;
return 48000;
}
#endif
case kDecoderCNGswb48kHz: {

86
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
View File

@ -607,7 +607,7 @@ class AudioDecoderCeltStereoTest : public AudioDecoderTest {
class AudioDecoderOpusTest : public AudioDecoderTest {
protected:
AudioDecoderOpusTest() : AudioDecoderTest() {
frame_size_ = 320;
frame_size_ = 480;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderOpus(kDecoderOpus);
assert(decoder_);
@ -618,75 +618,69 @@ class AudioDecoderOpusTest : public AudioDecoderTest {
WebRtcOpus_EncoderFree(encoder_);
}
virtual void SetUp() OVERRIDE {
AudioDecoderTest::SetUp();
// Upsample from 32 to 48 kHz.
// Because Opus is 48 kHz codec but the input file is 32 kHz, so the data
// read in |AudioDecoderTest::SetUp| has to be upsampled.
// |AudioDecoderTest::SetUp| has read |data_length_| samples, which is more
// than necessary after upsampling, so the end of audio that has been read
// is unused and the end of the buffer is overwritten by the resampled data.
Resampler rs;
rs.Reset(32000, 48000, kResamplerSynchronous);
const int before_resamp_len_samples = static_cast<int>(data_length_) * 2
/ 3;
int16_t* before_resamp_input = new int16_t[before_resamp_len_samples];
memcpy(before_resamp_input, input_,
sizeof(int16_t) * before_resamp_len_samples);
int resamp_len_samples;
EXPECT_EQ(0, rs.Push(before_resamp_input, before_resamp_len_samples,
input_, static_cast<int>(data_length_),
resamp_len_samples));
EXPECT_EQ(static_cast<int>(data_length_), resamp_len_samples);
delete[] before_resamp_input;
}
virtual void InitEncoder() {}
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
uint8_t* output) {
// Upsample from 32 to 48 kHz.
Resampler rs;
rs.Reset(32000, 48000, kResamplerSynchronous);
const int max_resamp_len_samples = static_cast<int>(input_len_samples) *
3 / 2;
int16_t* resamp_input = new int16_t[max_resamp_len_samples];
int resamp_len_samples;
EXPECT_EQ(0, rs.Push(input, static_cast<int>(input_len_samples),
resamp_input, max_resamp_len_samples,
resamp_len_samples));
EXPECT_EQ(max_resamp_len_samples, resamp_len_samples);
int enc_len_bytes =
WebRtcOpus_Encode(encoder_, resamp_input, resamp_len_samples,
static_cast<int>(data_length_), output);
uint8_t* output) OVERRIDE {
int enc_len_bytes = WebRtcOpus_Encode(encoder_, const_cast<int16_t*>(input),
static_cast<int16_t>(input_len_samples),
static_cast<int16_t>(data_length_), output);
EXPECT_GT(enc_len_bytes, 0);
delete [] resamp_input;
return enc_len_bytes;
}
OpusEncInst* encoder_;
};
class AudioDecoderOpusStereoTest : public AudioDecoderTest {
class AudioDecoderOpusStereoTest : public AudioDecoderOpusTest {
protected:
AudioDecoderOpusStereoTest() : AudioDecoderTest() {
AudioDecoderOpusStereoTest() : AudioDecoderOpusTest() {
channels_ = 2;
frame_size_ = 320;
data_length_ = 10 * frame_size_;
WebRtcOpus_EncoderFree(encoder_);
delete decoder_;
decoder_ = new AudioDecoderOpus(kDecoderOpus_2ch);
assert(decoder_);
WebRtcOpus_EncoderCreate(&encoder_, 2);
}
~AudioDecoderOpusStereoTest() {
WebRtcOpus_EncoderFree(encoder_);
}
virtual void InitEncoder() {}
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
uint8_t* output) {
uint8_t* output) OVERRIDE {
// Create stereo by duplicating each sample in |input|.
const int input_stereo_samples = static_cast<int>(input_len_samples) * 2;
int16_t* input_stereo = new int16_t[input_stereo_samples];
for (size_t i = 0; i < input_len_samples; i++)
input_stereo[i * 2] = input_stereo[i * 2 + 1] = input[i];
// Upsample from 32 to 48 kHz.
Resampler rs;
rs.Reset(32000, 48000, kResamplerSynchronousStereo);
const int max_resamp_len_samples = input_stereo_samples * 3 / 2;
int16_t* resamp_input = new int16_t[max_resamp_len_samples];
int resamp_len_samples;
EXPECT_EQ(0, rs.Push(input_stereo, input_stereo_samples, resamp_input,
max_resamp_len_samples, resamp_len_samples));
EXPECT_EQ(max_resamp_len_samples, resamp_len_samples);
int enc_len_bytes =
WebRtcOpus_Encode(encoder_, resamp_input, resamp_len_samples / 2,
static_cast<int16_t>(data_length_), output);
int enc_len_bytes = WebRtcOpus_Encode(
encoder_, input_stereo, static_cast<int16_t>(input_len_samples),
static_cast<int16_t>(data_length_), output);
EXPECT_GT(enc_len_bytes, 0);
delete [] resamp_input;
delete [] input_stereo;
delete[] input_stereo;
return enc_len_bytes;
}
OpusEncInst* encoder_;
};
TEST_F(AudioDecoderPcmUTest, EncodeDecode) {
@ -876,11 +870,11 @@ TEST(AudioDecoder, CodecSampleRateHz) {
EXPECT_EQ(8000, AudioDecoder::CodecSampleRateHz(kDecoderCNGnb));
EXPECT_EQ(16000, AudioDecoder::CodecSampleRateHz(kDecoderCNGwb));
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderCNGswb32kHz));
EXPECT_EQ(48000, AudioDecoder::CodecSampleRateHz(kDecoderOpus));
EXPECT_EQ(48000, AudioDecoder::CodecSampleRateHz(kDecoderOpus_2ch));
// TODO(tlegrand): Change 32000 to 48000 below once ACM has 48 kHz support.
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderCNGswb48kHz));
EXPECT_EQ(-1, AudioDecoder::CodecSampleRateHz(kDecoderArbitrary));
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderOpus));
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderOpus_2ch));
#ifdef WEBRTC_CODEC_CELT
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderCELT_32));
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderCELT_32_2ch));

2
webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
View File

@ -743,7 +743,7 @@ TEST(FecPayloadSplitter, MixedPayload) {
// Check first packet.
packet = packet_list.front();
EXPECT_EQ(0, packet->header.payloadType);
EXPECT_EQ(kBaseTimestamp - 20 * 32, packet->header.timestamp);
EXPECT_EQ(kBaseTimestamp - 20 * 48, packet->header.timestamp);
EXPECT_EQ(10, packet->payload_length);
EXPECT_FALSE(packet->primary);
delete [] packet->payload;

7
webrtc/modules/audio_coding/neteq/test/neteq_opus_fec_quality_test.cc
View File

@ -22,8 +22,7 @@ namespace webrtc {
namespace test {
static const int kOpusBlockDurationMs = 20;
static const int kOpusInputSamplingKhz = 48;
static const int kOpusOutputSamplingKhz = 32;
static const int kOpusSamplingKhz = 48;
static bool ValidateInFilename(const char* flagname, const string& value) {
FILE* fid = fopen(value.c_str(), "rb");
@ -117,8 +116,8 @@ class NetEqOpusFecQualityTest : public NetEqQualityTest {
};
NetEqOpusFecQualityTest::NetEqOpusFecQualityTest()
: NetEqQualityTest(kOpusBlockDurationMs, kOpusInputSamplingKhz,
kOpusOutputSamplingKhz,
: NetEqQualityTest(kOpusBlockDurationMs, kOpusSamplingKhz,
kOpusSamplingKhz,
(FLAGS_channels == 1) ? kDecoderOpus : kDecoderOpus_2ch,
FLAGS_channels, 0.0f, FLAGS_in_filename,
FLAGS_out_filename),

2
webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
View File

@ -48,8 +48,6 @@ uint32_t TimestampScaler::ToInternal(uint32_t external_timestamp,
denominator_ = 1;
break;
}
case kDecoderOpus:
case kDecoderOpus_2ch:
case kDecoderISACfb:
case kDecoderCNGswb48kHz: {
// Use timestamp scaling with factor 2/3 (32 kHz sample rate, but RTP

13
webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
View File

@ -252,10 +252,14 @@ TEST(TimestampScaler, TestG722Reset) {
EXPECT_CALL(db, Die()); // Called when database object is deleted.
}
// TODO(minyue): This test becomes trivial since Opus does not need a timestamp
// scaler. Therefore, this test may be removed in future. There is no harm to
// keep it, since it can be taken as a test case for the situation of a trivial
// timestamp scaler.
TEST(TimestampScaler, TestOpusLargeStep) {
MockDecoderDatabase db;
DecoderDatabase::DecoderInfo info;
info.codec_type = kDecoderOpus; // Uses a factor 2/3 scaling.
info.codec_type = kDecoderOpus;
static const uint8_t kRtpPayloadType = 17;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@ -273,8 +277,7 @@ TEST(TimestampScaler, TestOpusLargeStep) {
scaler.ToInternal(external_timestamp, kRtpPayloadType));
// Scale back.
EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
// Internal timestamp should be incremented with twice the step.
internal_timestamp += 2 * kStep / 3;
internal_timestamp += kStep;
}
EXPECT_CALL(db, Die()); // Called when database object is deleted.
@ -283,7 +286,7 @@ TEST(TimestampScaler, TestOpusLargeStep) {
TEST(TimestampScaler, TestIsacFbLargeStep) {
MockDecoderDatabase db;
DecoderDatabase::DecoderInfo info;
info.codec_type = kDecoderISACfb; // Uses a factor 2/3 scaling.
info.codec_type = kDecoderISACfb;
static const uint8_t kRtpPayloadType = 17;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@ -301,7 +304,7 @@ TEST(TimestampScaler, TestIsacFbLargeStep) {
scaler.ToInternal(external_timestamp, kRtpPayloadType));
// Scale back.
EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
// Internal timestamp should be incremented with twice the step.
// Internal timestamp should be incremented with two-thirds the step.
internal_timestamp += 2 * kStep / 3;
}