Reland of Opus multistream.
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/111750.
This time we don't use the multistream decoder unless we have to.
(Which is when #channels >2). Pros: don't make downstream projects
crash due to used up stack space, a few % more efficiency for the
typical case (because multistream adds some overhead). Cons: Messy
C-code with "union" types and #define MACROs, probably more
maintenance.
Bug: webrtc:8649
Change-Id: I4253a5e0c382f67ac7c6731dc6602a31e6779e63
Reviewed-on: https://webrtc-review.googlesource.com/c/120049
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26445}diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc
index e5f0464..de08827 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.
@@ -382,6 +386,11 @@
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
+#define ENCODER_CTL(inst, vargs) \
+ inst->channels <= 2 \
+ ? opus_encoder_ctl(inst->encoder.encoder, vargs) \
+ : opus_multistream_encoder_ctl(inst->encoder.multistream_encoder, vargs)
+
TEST_P(OpusTest, OpusEncodeDecode) {
PrepareSpeechData(channels_, 20, 20);
@@ -399,7 +408,7 @@
// Check application mode.
opus_int32 app;
- opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_APPLICATION(&app));
+ ENCODER_CTL(opus_encoder_, OPUS_GET_APPLICATION(&app));
EXPECT_EQ(application_ == 0 ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO,
app);
@@ -450,6 +459,11 @@
}
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 +509,7 @@
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 +582,17 @@
opus_int32 dtx;
// DTX is off by default.
- opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
+ ENCODER_CTL(opus_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));
+ ENCODER_CTL(opus_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));
+ ENCODER_CTL(opus_encoder_, OPUS_GET_DTX(&dtx));
EXPECT_EQ(0, dtx);
// Free memory.
@@ -592,6 +606,11 @@
}
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 +742,12 @@
}
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 +812,6 @@
INSTANTIATE_TEST_CASE_P(VariousMode,
OpusTest,
- Combine(Values(1, 2), Values(0, 1)));
+ Combine(Values(1, 2, 4), Values(0, 1)));
} // namespace webrtc