Opus multistream.
This is a backwards-compatible change. It makes WebRTC use the Opus
multistream decoder for all Opus packets. Single-stream packets are a
special case of multistream ones (with stream=1).
The tricky parts are 'WebRtcOpus_GetMaxPlaybackRate' and
'WebRtcOpus_GetSurroundParameters'. GetMaxPlaybackRate is supposed to
do what opus_encoder_ctl(encoder, OPUS_GET_MAX_BANDWIDTH(&bandwidth))
did when we had single-stream encoders. Now there may be several
independent encoders with possibly different BANDWIDTH. The new
GetMaxPlaybackRate queries all of them, and returns a playback rate if
all the encoder's rates are equal.
WebRtcOpus_GetSurroundParameters is a configuration convention. It
maps the number of channels to a multi-stream encoder/decoder
configuration. As described in RFC 7845
https://tools.ietf.org/html/rfc7845#section-5.1.1, a multi-stream
encoder/decoder needs a number of streams, number of coupled streams
and a 255-byte mapping array. The function GetSurroundParameters
computes all of these from the number of channels. [1, 2, 4, 6, 8]
channels are supported.
Bug: webrtc:8649
Change-Id: I271de8e387d738254d6aa53af7fcf8644a53edb5
Reviewed-on: https://webrtc-review.googlesource.com/c/111750
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26293}diff --git a/modules/BUILD.gn b/modules/BUILD.gn
index 5c399e4..40ea128 100644
--- a/modules/BUILD.gn
+++ b/modules/BUILD.gn
@@ -124,6 +124,7 @@
"../resources/audio_coding/neteq_universal_new.rtp",
"../resources/audio_coding/speech_mono_16kHz.pcm",
"../resources/audio_coding/speech_mono_32_48kHz.pcm",
+ "../resources/audio_coding/speech_4_channels_48k_one_second.wav",
"../resources/audio_coding/testfile32kHz.pcm",
"../resources/audio_coding/teststereo32kHz.pcm",
"../resources/audio_device/audio_short16.pcm",
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index 357cb1a..1accfe4 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -72,7 +72,8 @@
AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels)
: channels_(num_channels) {
RTC_DCHECK(num_channels == 1 || num_channels == 2);
- WebRtcOpus_DecoderCreate(&dec_state_, channels_);
+ const int error = WebRtcOpus_DecoderCreate(&dec_state_, channels_);
+ RTC_DCHECK(error == 0);
WebRtcOpus_DecoderInit(dec_state_);
}
diff --git a/modules/audio_coding/codecs/opus/opus_inst.h b/modules/audio_coding/codecs/opus/opus_inst.h
index 2473a5c..41b3f15 100644
--- a/modules/audio_coding/codecs/opus/opus_inst.h
+++ b/modules/audio_coding/codecs/opus/opus_inst.h
@@ -17,16 +17,17 @@
RTC_PUSH_IGNORING_WUNDEF()
#include "opus.h"
+#include "opus_multistream.h"
RTC_POP_IGNORING_WUNDEF()
struct WebRtcOpusEncInst {
- OpusEncoder* encoder;
+ OpusMSEncoder* encoder;
size_t channels;
int in_dtx_mode;
};
struct WebRtcOpusDecInst {
- OpusDecoder* decoder;
+ OpusMSDecoder* decoder;
int prev_decoded_samples;
size_t channels;
int in_dtx_mode;
diff --git a/modules/audio_coding/codecs/opus/opus_interface.c b/modules/audio_coding/codecs/opus/opus_interface.c
index d219098..c657a14 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/modules/audio_coding/codecs/opus/opus_interface.c
@@ -37,6 +37,40 @@
kWebRtcOpusDefaultFrameSize = 960,
};
+int16_t GetSurroundParameters(int channels,
+ int *streams,
+ int *coupled_streams,
+ unsigned char *mapping) {
+ int opus_error;
+ int ret = 0;
+ // Use 'surround encoder create' to get values for 'coupled_streams',
+ // 'streams' and 'mapping'.
+ OpusMSEncoder* ms_encoder_ptr = opus_multistream_surround_encoder_create(
+ 48000,
+ channels,
+ /* mapping family */ channels <= 2 ? 0 : 1,
+ streams,
+ coupled_streams,
+ mapping,
+ OPUS_APPLICATION_VOIP, // Application type shouldn't affect
+ // streams/mapping values.
+ &opus_error);
+
+ // This shouldn't fail; if it fails,
+ // signal an error and return invalid values.
+ if (opus_error != OPUS_OK || ms_encoder_ptr == NULL) {
+ ret = -1;
+ *streams = -1;
+ *coupled_streams = -1;
+ }
+
+ // We don't need the encoder.
+ if (ms_encoder_ptr != NULL) {
+ opus_multistream_encoder_destroy(ms_encoder_ptr);
+ }
+ return ret;
+}
+
int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
size_t channels,
int32_t application) {
@@ -55,12 +89,26 @@
return -1;
}
+ unsigned char mapping[255];
+ memset(mapping, 0, 255);
+ int streams = -1;
+ int coupled_streams = -1;
+
+
OpusEncInst* state = calloc(1, sizeof(OpusEncInst));
RTC_DCHECK(state);
int error;
- state->encoder = opus_encoder_create(48000, (int)channels, opus_app,
- &error);
+ state->encoder = opus_multistream_surround_encoder_create(
+ 48000,
+ channels,
+ /* mapping family */ channels <= 2 ? 0 : 1,
+ &streams,
+ &coupled_streams,
+ mapping,
+ opus_app,
+ &error);
+
if (error != OPUS_OK || !state->encoder) {
WebRtcOpus_EncoderFree(state);
return -1;
@@ -75,7 +123,7 @@
int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
if (inst) {
- opus_encoder_destroy(inst->encoder);
+ opus_multistream_encoder_destroy(inst->encoder);
free(inst);
return 0;
} else {
@@ -94,11 +142,11 @@
return -1;
}
- res = opus_encode(inst->encoder,
- (const opus_int16*)audio_in,
- (int)samples,
- encoded,
- (opus_int32)length_encoded_buffer);
+ res = opus_multistream_encode(inst->encoder,
+ (const opus_int16*)audio_in,
+ (int)samples,
+ encoded,
+ (opus_int32)length_encoded_buffer);
if (res <= 0) {
return -1;
@@ -122,7 +170,7 @@
int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate) {
if (inst) {
- return opus_encoder_ctl(inst->encoder, OPUS_SET_BITRATE(rate));
+ return opus_multistream_encoder_ctl(inst->encoder, OPUS_SET_BITRATE(rate));
} else {
return -1;
}
@@ -130,8 +178,8 @@
int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate) {
if (inst) {
- return opus_encoder_ctl(inst->encoder,
- OPUS_SET_PACKET_LOSS_PERC(loss_rate));
+ return opus_multistream_encoder_ctl(inst->encoder,
+ OPUS_SET_PACKET_LOSS_PERC(loss_rate));
} else {
return -1;
}
@@ -154,13 +202,46 @@
} else {
set_bandwidth = OPUS_BANDWIDTH_FULLBAND;
}
- return opus_encoder_ctl(inst->encoder,
- OPUS_SET_MAX_BANDWIDTH(set_bandwidth));
+ return opus_multistream_encoder_ctl(inst->encoder,
+ OPUS_SET_MAX_BANDWIDTH(set_bandwidth));
+}
+
+int16_t WebRtcOpus_GetMaxPlaybackRate(OpusEncInst* const inst,
+ int32_t* result_hz) {
+ opus_int32 max_bandwidth;
+ int s;
+ int ret;
+
+ max_bandwidth = 0;
+ ret = OPUS_OK;
+ s = 0;
+ while (ret == OPUS_OK) {
+ OpusEncoder *enc;
+ opus_int32 bandwidth;
+
+ ret = opus_multistream_encoder_ctl(
+ inst->encoder,
+ OPUS_MULTISTREAM_GET_ENCODER_STATE(s, &enc));
+ if (ret == OPUS_BAD_ARG)
+ break;
+ if (ret != OPUS_OK)
+ return -1;
+ if (opus_encoder_ctl(enc, OPUS_GET_MAX_BANDWIDTH(&bandwidth)) != OPUS_OK)
+ return -1;
+
+ if (max_bandwidth != 0 && max_bandwidth != bandwidth)
+ return -1;
+
+ max_bandwidth = bandwidth;
+ s++;
+ }
+ *result_hz = max_bandwidth;
+ return 0;
}
int16_t WebRtcOpus_EnableFec(OpusEncInst* inst) {
if (inst) {
- return opus_encoder_ctl(inst->encoder, OPUS_SET_INBAND_FEC(1));
+ return opus_multistream_encoder_ctl(inst->encoder, OPUS_SET_INBAND_FEC(1));
} else {
return -1;
}
@@ -168,7 +249,7 @@
int16_t WebRtcOpus_DisableFec(OpusEncInst* inst) {
if (inst) {
- return opus_encoder_ctl(inst->encoder, OPUS_SET_INBAND_FEC(0));
+ return opus_multistream_encoder_ctl(inst->encoder, OPUS_SET_INBAND_FEC(0));
} else {
return -1;
}
@@ -184,21 +265,21 @@
// last long during a pure silence, if the signal type is not forced.
// TODO(minyue): Remove the signal type forcing when Opus DTX works properly
// without it.
- int ret = opus_encoder_ctl(inst->encoder,
- OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE));
+ int ret = opus_multistream_encoder_ctl(inst->encoder,
+ OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE));
if (ret != OPUS_OK)
return ret;
- return opus_encoder_ctl(inst->encoder, OPUS_SET_DTX(1));
+ return opus_multistream_encoder_ctl(inst->encoder, OPUS_SET_DTX(1));
}
int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst) {
if (inst) {
- int ret = opus_encoder_ctl(inst->encoder,
- OPUS_SET_SIGNAL(OPUS_AUTO));
+ int ret = opus_multistream_encoder_ctl(inst->encoder,
+ OPUS_SET_SIGNAL(OPUS_AUTO));
if (ret != OPUS_OK)
return ret;
- return opus_encoder_ctl(inst->encoder, OPUS_SET_DTX(0));
+ return opus_multistream_encoder_ctl(inst->encoder, OPUS_SET_DTX(0));
} else {
return -1;
}
@@ -206,7 +287,7 @@
int16_t WebRtcOpus_EnableCbr(OpusEncInst* inst) {
if (inst) {
- return opus_encoder_ctl(inst->encoder, OPUS_SET_VBR(0));
+ return opus_multistream_encoder_ctl(inst->encoder, OPUS_SET_VBR(0));
} else {
return -1;
}
@@ -214,7 +295,7 @@
int16_t WebRtcOpus_DisableCbr(OpusEncInst* inst) {
if (inst) {
- return opus_encoder_ctl(inst->encoder, OPUS_SET_VBR(1));
+ return opus_multistream_encoder_ctl(inst->encoder, OPUS_SET_VBR(1));
} else {
return -1;
}
@@ -222,7 +303,8 @@
int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) {
if (inst) {
- return opus_encoder_ctl(inst->encoder, OPUS_SET_COMPLEXITY(complexity));
+ return opus_multistream_encoder_ctl(inst->encoder,
+ OPUS_SET_COMPLEXITY(complexity));
} else {
return -1;
}
@@ -233,7 +315,8 @@
return -1;
}
int32_t bandwidth;
- if (opus_encoder_ctl(inst->encoder, OPUS_GET_BANDWIDTH(&bandwidth)) == 0) {
+ if (opus_multistream_encoder_ctl(inst->encoder,
+ OPUS_GET_BANDWIDTH(&bandwidth)) == 0) {
return bandwidth;
} else {
return -1;
@@ -243,7 +326,8 @@
int16_t WebRtcOpus_SetBandwidth(OpusEncInst* inst, int32_t bandwidth) {
if (inst) {
- return opus_encoder_ctl(inst->encoder, OPUS_SET_BANDWIDTH(bandwidth));
+ return opus_multistream_encoder_ctl(inst->encoder,
+ OPUS_SET_BANDWIDTH(bandwidth));
} else {
return -1;
}
@@ -253,10 +337,10 @@
if (!inst)
return -1;
if (num_channels == 0) {
- return opus_encoder_ctl(inst->encoder,
+ return opus_multistream_encoder_ctl(inst->encoder,
OPUS_SET_FORCE_CHANNELS(OPUS_AUTO));
} else if (num_channels == 1 || num_channels == 2) {
- return opus_encoder_ctl(inst->encoder,
+ return opus_multistream_encoder_ctl(inst->encoder,
OPUS_SET_FORCE_CHANNELS(num_channels));
} else {
return -1;
@@ -268,16 +352,31 @@
OpusDecInst* state;
if (inst != NULL) {
- /* Create Opus decoder state. */
+ // Create Opus decoder state.
state = (OpusDecInst*) calloc(1, sizeof(OpusDecInst));
if (state == NULL) {
return -1;
}
- /* Create new memory, always at 48000 Hz. */
- state->decoder = opus_decoder_create(48000, (int)channels, &error);
+ unsigned char mapping[255];
+ memset(mapping, 0, 255);
+ int streams = -1;
+ int coupled_streams = -1;
+ if (GetSurroundParameters(channels, &streams,
+ &coupled_streams, mapping) != 0) {
+ free(state);
+ return -1;
+ }
+
+ // Create new memory, always at 48000 Hz.
+ state->decoder = opus_multistream_decoder_create(
+ 48000, (int)channels,
+ /* streams = */ streams,
+ /* coupled streams = */ coupled_streams,
+ mapping,
+ &error);
if (error == OPUS_OK && state->decoder != NULL) {
- /* Creation of memory all ok. */
+ // Creation of memory all ok.
state->channels = channels;
state->prev_decoded_samples = kWebRtcOpusDefaultFrameSize;
state->in_dtx_mode = 0;
@@ -285,9 +384,9 @@
return 0;
}
- /* If memory allocation was unsuccessful, free the entire state. */
+ // If memory allocation was unsuccessful, free the entire state.
if (state->decoder) {
- opus_decoder_destroy(state->decoder);
+ opus_multistream_decoder_destroy(state->decoder);
}
free(state);
}
@@ -296,7 +395,7 @@
int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst) {
if (inst) {
- opus_decoder_destroy(inst->decoder);
+ opus_multistream_decoder_destroy(inst->decoder);
free(inst);
return 0;
} else {
@@ -309,7 +408,7 @@
}
void WebRtcOpus_DecoderInit(OpusDecInst* inst) {
- opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
+ opus_multistream_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
inst->in_dtx_mode = 0;
}
@@ -324,6 +423,10 @@
// fact a 1-byte TOC with a 1-byte payload. That will be erroneously
// interpreted as comfort noise output, but such a payload is probably
// faulty anyway.
+
+ // TODO(webrtc:10218): This is wrong for multistream opus. Then are several
+ // single-stream packets glued together with some packet size bytes in
+ // between. See https://tools.ietf.org/html/rfc6716#appendix-B
inst->in_dtx_mode = 1;
return 2; // Comfort noise.
} else {
@@ -338,8 +441,9 @@
static int DecodeNative(OpusDecInst* inst, const uint8_t* encoded,
size_t encoded_bytes, int frame_size,
int16_t* decoded, int16_t* audio_type, int decode_fec) {
- int res = opus_decode(inst->decoder, encoded, (opus_int32)encoded_bytes,
- (opus_int16*)decoded, frame_size, decode_fec);
+ int res = opus_multistream_decode(
+ inst->decoder, encoded, (opus_int32)encoded_bytes,
+ (opus_int16*)decoded, frame_size, decode_fec);
if (res <= 0)
return -1;
diff --git a/modules/audio_coding/codecs/opus/opus_interface.h b/modules/audio_coding/codecs/opus/opus_interface.h
index ddb4ff9..0e97734 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.h
+++ b/modules/audio_coding/codecs/opus/opus_interface.h
@@ -125,6 +125,22 @@
*/
int16_t WebRtcOpus_SetMaxPlaybackRate(OpusEncInst* inst, int32_t frequency_hz);
+/****************************************************************************
+ * WebRtcOpus_GetMaxPlaybackRate(...)
+ *
+ * Queries the maximum playback rate for encoding. If different single-stream
+ * encoders have different maximum playback rates, this function fails.
+ *
+ * Input:
+ * - inst : Encoder context.
+ * Output:
+ * - result_hz : The maximum playback rate in Hz.
+ * Return value : 0 - Success
+ * -1 - Error
+ */
+int16_t WebRtcOpus_GetMaxPlaybackRate(OpusEncInst* const inst,
+ int32_t* result_hz);
+
/* TODO(minyue): Check whether an API to check the FEC and the packet loss rate
* is needed. It might not be very useful since there are not many use cases and
* the caller can always maintain the states. */
diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc
index e5f0464..50178a9 100644
--- a/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -27,7 +27,7 @@
using ::testing::Combine;
// Maximum number of bytes in output bitstream.
-const size_t kMaxBytes = 1000;
+const size_t kMaxBytes = 2000;
// Sample rate of Opus.
const size_t kOpusRateKhz = 48;
// Number of samples-per-channel in a 20 ms frame, sampled at 48 kHz.
@@ -86,10 +86,14 @@
void OpusTest::PrepareSpeechData(size_t channel,
int block_length_ms,
int loop_length_ms) {
+ std::map<int, std::string> channel_to_basename = {
+ {1, "audio_coding/testfile32kHz"},
+ {2, "audio_coding/teststereo32kHz"},
+ {4, "audio_coding/speech_4_channels_48k_one_second"}};
+ std::map<int, std::string> channel_to_suffix = {
+ {1, "pcm"}, {2, "pcm"}, {4, "wav"}};
const std::string file_name = webrtc::test::ResourcePath(
- (channel == 1) ? "audio_coding/testfile32kHz"
- : "audio_coding/teststereo32kHz",
- "pcm");
+ channel_to_basename[channel], channel_to_suffix[channel]);
if (loop_length_ms < block_length_ms) {
loop_length_ms = block_length_ms;
}
@@ -103,7 +107,7 @@
int32_t set) {
opus_int32 bandwidth;
EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, set));
- opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_MAX_BANDWIDTH(&bandwidth));
+ EXPECT_EQ(0, WebRtcOpus_GetMaxPlaybackRate(opus_encoder_, &bandwidth));
EXPECT_EQ(expect, bandwidth);
}
@@ -354,13 +358,13 @@
// Test to see that an invalid pointer is caught.
EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(NULL, 1, 0));
// Invalid channel number.
- EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 3, 0));
+ EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 257, 0));
// Invalid applciation mode.
EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 1, 2));
EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(NULL, 1));
// Invalid channel number.
- EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 3));
+ EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 257));
}
// Test failing Free.
@@ -399,7 +403,8 @@
// Check application mode.
opus_int32 app;
- opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_APPLICATION(&app));
+ opus_multistream_encoder_ctl(opus_encoder_->encoder,
+ OPUS_GET_APPLICATION(&app));
EXPECT_EQ(application_ == 0 ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO,
app);
@@ -450,6 +455,11 @@
}
TEST_P(OpusTest, OpusSetBandwidth) {
+ if (channels_ > 2) {
+ // TODO(webrtc:10217): investigate why multi-stream Opus reports
+ // narrowband when it's configured with FULLBAND.
+ return;
+ }
PrepareSpeechData(channels_, 20, 20);
int16_t audio_type;
@@ -495,7 +505,7 @@
ASSERT_EQ(0,
WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
- if (channels_ == 2) {
+ if (channels_ >= 2) {
EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 3));
EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 2));
EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 1));
@@ -568,17 +578,17 @@
opus_int32 dtx;
// DTX is off by default.
- opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
+ opus_multistream_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
EXPECT_EQ(0, dtx);
// Test to enable DTX.
EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_));
- opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
+ opus_multistream_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
EXPECT_EQ(1, dtx);
// Test to disable DTX.
EXPECT_EQ(0, WebRtcOpus_DisableDtx(opus_encoder_));
- opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
+ opus_multistream_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
EXPECT_EQ(0, dtx);
// Free memory.
@@ -592,6 +602,11 @@
}
TEST_P(OpusTest, OpusDtxOn) {
+ if (channels_ > 2) {
+ // TODO(webrtc:10218): adapt the test to the sizes and order of multi-stream
+ // DTX packets.
+ return;
+ }
TestDtxEffect(true, 10);
TestDtxEffect(true, 20);
TestDtxEffect(true, 40);
@@ -723,6 +738,12 @@
}
TEST_P(OpusTest, OpusDecodeRepacketized) {
+ if (channels_ > 2) {
+ // As per the Opus documentation
+ // https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__repacketizer.html#details,
+ // multiple streams are not supported.
+ return;
+ }
constexpr size_t kPackets = 6;
PrepareSpeechData(channels_, 20, 20 * kPackets);
@@ -787,6 +808,6 @@
INSTANTIATE_TEST_CASE_P(VariousMode,
OpusTest,
- Combine(Values(1, 2), Values(0, 1)));
+ Combine(Values(1, 2, 4), Values(0, 1)));
} // namespace webrtc
diff --git a/resources/audio_coding/speech_4_channels_48k_one_second.wav.sha1 b/resources/audio_coding/speech_4_channels_48k_one_second.wav.sha1
new file mode 100644
index 0000000..7d3041c
--- /dev/null
+++ b/resources/audio_coding/speech_4_channels_48k_one_second.wav.sha1
@@ -0,0 +1 @@
+a60c7d03ac2ad9af3cfc7640a4979881f6d47c9c
\ No newline at end of file
