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}

modules/BUILD.gn

@@ -124,6 +124,7 @@ if (rtc_include_tests) {
     "../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",

modules/audio_coding/codecs/opus/audio_decoder_opus.cc

@@ -72,7 +72,8 @@ class OpusFrame : public AudioDecoder::EncodedAudioFrame {
 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_);
 }
 

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;

modules/audio_coding/codecs/opus/opus_interface.c

@@ -37,6 +37,40 @@ enum {
   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 @@ int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
       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,
+  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_EncoderCreate(OpusEncInst** inst,
 
 int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
   if (inst) {
-    opus_encoder_destroy(inst->encoder);
+    opus_multistream_encoder_destroy(inst->encoder);
     free(inst);
     return 0;
   } else {
@@ -94,7 +142,7 @@ int WebRtcOpus_Encode(OpusEncInst* inst,
     return -1;
   }
 
-  res = opus_encode(inst->encoder,
+  res = opus_multistream_encode(inst->encoder,
                                 (const opus_int16*)audio_in,
                                 (int)samples,
                                 encoded,
@@ -122,7 +170,7 @@ int WebRtcOpus_Encode(OpusEncInst* inst,
 
 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;
   }
@@ -130,7 +178,7 @@ int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate) {
 
 int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate) {
   if (inst) {
-    return opus_encoder_ctl(inst->encoder,
+    return opus_multistream_encoder_ctl(inst->encoder,
                                         OPUS_SET_PACKET_LOSS_PERC(loss_rate));
   } else {
     return -1;
@@ -154,13 +202,46 @@ int16_t WebRtcOpus_SetMaxPlaybackRate(OpusEncInst* inst, int32_t frequency_hz) {
   } else {
     set_bandwidth = OPUS_BANDWIDTH_FULLBAND;
   }
-  return opus_encoder_ctl(inst->encoder,
+  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_EnableFec(OpusEncInst* inst) {
 
 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 @@ int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst) {
   // 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,
+  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,
+    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_DisableDtx(OpusEncInst* inst) {
 
 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_EnableCbr(OpusEncInst* inst) {
 
 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_DisableCbr(OpusEncInst* inst) {
 
 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 @@ int32_t WebRtcOpus_GetBandwidth(OpusEncInst* inst) {
     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 @@ int32_t WebRtcOpus_GetBandwidth(OpusEncInst* inst) {
 
 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 @@ int16_t WebRtcOpus_SetForceChannels(OpusEncInst* inst, size_t num_channels) {
   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 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, size_t channels) {
   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. */
+    unsigned char mapping[255];
-    state->decoder = opus_decoder_create(48000, (int)channels, &error);
+    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 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, size_t channels) {
       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_DecoderCreate(OpusDecInst** inst, size_t channels) {
 
 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 @@ size_t WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
 }
 
 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 @@ static int16_t DetermineAudioType(OpusDecInst* inst, size_t encoded_bytes) {
     // 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 {
@@ -338,7 +441,8 @@ static int16_t DetermineAudioType(OpusDecInst* inst, size_t encoded_bytes) {
 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,
+  int res = opus_multistream_decode(
+      inst->decoder, encoded, (opus_int32)encoded_bytes,
       (opus_int16*)decoded, frame_size, decode_fec);
 
   if (res <= 0)

modules/audio_coding/codecs/opus/opus_interface.h

@@ -125,6 +125,22 @@ int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate);
  */
 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. */

modules/audio_coding/codecs/opus/opus_unittest.cc

@@ -27,7 +27,7 @@ using ::testing::Values;
 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 @@ OpusTest::OpusTest()
 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"
+      channel_to_basename[channel], channel_to_suffix[channel]);
-                     : "audio_coding/teststereo32kHz",
-      "pcm");
   if (loop_length_ms < block_length_ms) {
     loop_length_ms = block_length_ms;
   }
@@ -103,7 +107,7 @@ void OpusTest::SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
                                   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(OpusTest, OpusCreateFail) {
   // 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 @@ TEST_P(OpusTest, OpusEncodeDecode) {
 
   // 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, OpusSetComplexity) {
 }
 
 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 @@ TEST_P(OpusTest, OpusForceChannels) {
   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 @@ TEST_P(OpusTest, OpusEnableDisableDtx) {
   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, OpusDtxOff) {
 }
 
 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, OpusDurationEstimation) {
 }
 
 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 @@ TEST_P(OpusTest, OpusDecodeRepacketized) {
 
 INSTANTIATE_TEST_CASE_P(VariousMode,
                         OpusTest,
-                        Combine(Values(1, 2), Values(0, 1)));
+                        Combine(Values(1, 2, 4), Values(0, 1)));
 
 }  // namespace webrtc

resources/audio_coding/speech_4_channels_48k_one_second.wav.sha1

@@ -0,0 +1 @@
+a60c7d03ac2ad9af3cfc7640a4979881f6d47c9c