Revert "Opus multistream."
This reverts commit 83ed89a45f4578ca07efef48e772b9aafb263163.
Reason for revert: breaks downstream project
Original change's description:
> 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}
TBR=aleloi@webrtc.org,minyue@webrtc.org
Change-Id: I1002e3273b44d3cccacdba84b8c363eefd537c4b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/118201
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26306}diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc
index 50178a9..e5f0464 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 = 2000;
+const size_t kMaxBytes = 1000;
// Sample rate of Opus.
const size_t kOpusRateKhz = 48;
// Number of samples-per-channel in a 20 ms frame, sampled at 48 kHz.
@@ -86,14 +86,10 @@
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_to_basename[channel], channel_to_suffix[channel]);
+ (channel == 1) ? "audio_coding/testfile32kHz"
+ : "audio_coding/teststereo32kHz",
+ "pcm");
if (loop_length_ms < block_length_ms) {
loop_length_ms = block_length_ms;
}
@@ -107,7 +103,7 @@
int32_t set) {
opus_int32 bandwidth;
EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, set));
- EXPECT_EQ(0, WebRtcOpus_GetMaxPlaybackRate(opus_encoder_, &bandwidth));
+ opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_MAX_BANDWIDTH(&bandwidth));
EXPECT_EQ(expect, bandwidth);
}
@@ -358,13 +354,13 @@
// 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, 257, 0));
+ EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 3, 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, 257));
+ EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 3));
}
// Test failing Free.
@@ -403,8 +399,7 @@
// Check application mode.
opus_int32 app;
- opus_multistream_encoder_ctl(opus_encoder_->encoder,
- OPUS_GET_APPLICATION(&app));
+ opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_APPLICATION(&app));
EXPECT_EQ(application_ == 0 ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO,
app);
@@ -455,11 +450,6 @@
}
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;
@@ -505,7 +495,7 @@
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));
@@ -578,17 +568,17 @@
opus_int32 dtx;
// DTX is off by default.
- opus_multistream_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
+ opus_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_multistream_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
+ opus_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_multistream_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
+ opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
EXPECT_EQ(0, dtx);
// Free memory.
@@ -602,11 +592,6 @@
}
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);
@@ -738,12 +723,6 @@
}
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);
@@ -808,6 +787,6 @@
INSTANTIATE_TEST_CASE_P(VariousMode,
OpusTest,
- Combine(Values(1, 2, 4), Values(0, 1)));
+ Combine(Values(1, 2), Values(0, 1)));
} // namespace webrtc