WebRTC Opus C interface: Add support for non-48 kHz decode sample rate
Plus tests for 16 kHz.
Bug: webrtc:10631
Change-Id: I2d89bc6d0d9548f0ad7bb1e36d6dfde6b6b31f83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138072
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28099}
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc
index df0f448..c8fd176 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc
+++ b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc
@@ -157,7 +157,7 @@
return PacketDuration(encoded, encoded_len);
}
- return WebRtcOpus_FecDurationEst(encoded, encoded_len);
+ return WebRtcOpus_FecDurationEst(encoded, encoded_len, 48000);
}
bool AudioDecoderMultiChannelOpusImpl::PacketHasFec(const uint8_t* encoded,
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index 77e1535..0369549 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -23,7 +23,7 @@
AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels)
: channels_(num_channels) {
RTC_DCHECK(num_channels == 1 || num_channels == 2);
- const int error = WebRtcOpus_DecoderCreate(&dec_state_, channels_);
+ const int error = WebRtcOpus_DecoderCreate(&dec_state_, channels_, 48000);
RTC_DCHECK(error == 0);
WebRtcOpus_DecoderInit(dec_state_);
}
@@ -104,7 +104,7 @@
return PacketDuration(encoded, encoded_len);
}
- return WebRtcOpus_FecDurationEst(encoded, encoded_len);
+ return WebRtcOpus_FecDurationEst(encoded, encoded_len, 48000);
}
bool AudioDecoderOpusImpl::PacketHasFec(const uint8_t* encoded,
diff --git a/modules/audio_coding/codecs/opus/opus_fec_test.cc b/modules/audio_coding/codecs/opus/opus_fec_test.cc
index 0af90fe..287213c 100644
--- a/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -110,7 +110,7 @@
// Create encoder memory.
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app, 48000));
- EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
+ EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_, 48000));
// Set bitrate.
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_));
}
diff --git a/modules/audio_coding/codecs/opus/opus_inst.h b/modules/audio_coding/codecs/opus/opus_inst.h
index 08c1e0f..9c3acb3 100644
--- a/modules/audio_coding/codecs/opus/opus_inst.h
+++ b/modules/audio_coding/codecs/opus/opus_inst.h
@@ -33,6 +33,7 @@
int prev_decoded_samples;
size_t channels;
int in_dtx_mode;
+ int sample_rate_hz;
};
#endif // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
diff --git a/modules/audio_coding/codecs/opus/opus_interface.c b/modules/audio_coding/codecs/opus/opus_interface.c
index 817caac..eb426c3 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/modules/audio_coding/codecs/opus/opus_interface.c
@@ -28,15 +28,26 @@
* side, we must allow for packets of that size. NetEq is currently limited
* to 60 ms on the receive side. */
kWebRtcOpusMaxDecodeFrameSizeMs = 120,
-
- /* Maximum sample count per channel is 48 kHz * maximum frame size in
- * milliseconds. */
- kWebRtcOpusMaxFrameSizePerChannel = 48 * kWebRtcOpusMaxDecodeFrameSizeMs,
-
- /* Default frame size, 20 ms @ 48 kHz, in samples (for one channel). */
- kWebRtcOpusDefaultFrameSize = 960,
};
+static int FrameSizePerChannel(int frame_size_ms, int sample_rate_hz) {
+ RTC_DCHECK_GT(frame_size_ms, 0);
+ RTC_DCHECK_EQ(frame_size_ms % 10, 0);
+ RTC_DCHECK_GT(sample_rate_hz, 0);
+ RTC_DCHECK_EQ(sample_rate_hz % 1000, 0);
+ return frame_size_ms * (sample_rate_hz / 1000);
+}
+
+// Maximum sample count per channel.
+static int MaxFrameSizePerChannel(int sample_rate_hz) {
+ return FrameSizePerChannel(kWebRtcOpusMaxDecodeFrameSizeMs, sample_rate_hz);
+}
+
+// Default sample count per channel.
+static int DefaultFrameSizePerChannel(int sample_rate_hz) {
+ return FrameSizePerChannel(20, sample_rate_hz);
+}
+
int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
size_t channels,
int32_t application,
@@ -374,7 +385,9 @@
}
}
-int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, size_t channels) {
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst,
+ size_t channels,
+ int sample_rate_hz) {
int error;
OpusDecInst* state;
@@ -385,14 +398,13 @@
return -1;
}
- // Create new memory, always at 48000 Hz.
- state->decoder = opus_decoder_create(48000,
- (int)channels, &error);
+ state->decoder = opus_decoder_create(sample_rate_hz, (int)channels, &error);
if (error == OPUS_OK && state->decoder) {
// Creation of memory all ok.
state->channels = channels;
- state->prev_decoded_samples = kWebRtcOpusDefaultFrameSize;
+ state->prev_decoded_samples = DefaultFrameSizePerChannel(sample_rate_hz);
state->in_dtx_mode = 0;
+ state->sample_rate_hz = sample_rate_hz;
*inst = state;
return 0;
}
@@ -432,8 +444,9 @@
if (error == OPUS_OK && state->multistream_decoder) {
// Creation of memory all ok.
state->channels = channels;
- state->prev_decoded_samples = kWebRtcOpusDefaultFrameSize;
+ state->prev_decoded_samples = DefaultFrameSizePerChannel(48000);
state->in_dtx_mode = 0;
+ state->sample_rate_hz = 48000;
*inst = state;
return 0;
}
@@ -529,13 +542,9 @@
*audio_type = DetermineAudioType(inst, encoded_bytes);
decoded_samples = WebRtcOpus_DecodePlc(inst, decoded, 1);
} else {
- decoded_samples = DecodeNative(inst,
- encoded,
- encoded_bytes,
- kWebRtcOpusMaxFrameSizePerChannel,
- decoded,
- audio_type,
- 0);
+ decoded_samples = DecodeNative(inst, encoded, encoded_bytes,
+ MaxFrameSizePerChannel(inst->sample_rate_hz),
+ decoded, audio_type, 0);
}
if (decoded_samples < 0) {
return -1;
@@ -555,10 +564,13 @@
/* The number of samples we ask for is |number_of_lost_frames| times
* |prev_decoded_samples_|. Limit the number of samples to maximum
- * |kWebRtcOpusMaxFrameSizePerChannel|. */
+ * |MaxFrameSizePerChannel()|. */
plc_samples = number_of_lost_frames * inst->prev_decoded_samples;
- plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ?
- plc_samples : kWebRtcOpusMaxFrameSizePerChannel;
+ const int max_samples_per_channel =
+ MaxFrameSizePerChannel(inst->sample_rate_hz);
+ plc_samples = plc_samples <= max_samples_per_channel
+ ? plc_samples
+ : max_samples_per_channel;
decoded_samples = DecodeNative(inst, NULL, 0, plc_samples,
decoded, &audio_type, 0);
if (decoded_samples < 0) {
@@ -578,7 +590,8 @@
return 0;
}
- fec_samples = opus_packet_get_samples_per_frame(encoded, 48000);
+ fec_samples =
+ opus_packet_get_samples_per_frame(encoded, inst->sample_rate_hz);
decoded_samples = DecodeNative(inst, encoded, encoded_bytes,
fec_samples, decoded, audio_type, 1);
@@ -604,9 +617,10 @@
/* Invalid payload data. */
return 0;
}
- samples = frames * opus_packet_get_samples_per_frame(payload, 48000);
- if (samples < 120 || samples > 5760) {
- /* Invalid payload duration. */
+ samples =
+ frames * opus_packet_get_samples_per_frame(payload, inst->sample_rate_hz);
+ if (samples > 120 * inst->sample_rate_hz / 1000) {
+ // More than 120 ms' worth of samples.
return 0;
}
return samples;
@@ -615,21 +629,24 @@
int WebRtcOpus_PlcDuration(OpusDecInst* inst) {
/* The number of samples we ask for is |number_of_lost_frames| times
* |prev_decoded_samples_|. Limit the number of samples to maximum
- * |kWebRtcOpusMaxFrameSizePerChannel|. */
+ * |MaxFrameSizePerChannel()|. */
const int plc_samples = inst->prev_decoded_samples;
- return (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ?
- plc_samples : kWebRtcOpusMaxFrameSizePerChannel;
+ const int max_samples_per_channel =
+ MaxFrameSizePerChannel(inst->sample_rate_hz);
+ return plc_samples <= max_samples_per_channel ? plc_samples
+ : max_samples_per_channel;
}
int WebRtcOpus_FecDurationEst(const uint8_t* payload,
- size_t payload_length_bytes) {
- int samples;
+ size_t payload_length_bytes,
+ int sample_rate_hz) {
if (WebRtcOpus_PacketHasFec(payload, payload_length_bytes) != 1) {
return 0;
}
-
- samples = opus_packet_get_samples_per_frame(payload, 48000);
- if (samples < 480 || samples > 5760) {
+ const int samples =
+ opus_packet_get_samples_per_frame(payload, sample_rate_hz);
+ const int samples_per_ms = sample_rate_hz / 1000;
+ if (samples < 10 * samples_per_ms || samples > 120 * samples_per_ms) {
/* Invalid payload duration. */
return 0;
}
@@ -650,6 +667,8 @@
if (payload[0] & 0x80)
return 0;
+ // For computing the payload length in ms, the sample rate is not important
+ // since it cancels out. We use 48 kHz, but any valid sample rate would work.
payload_length_ms = opus_packet_get_samples_per_frame(payload, 48000) / 48;
if (10 > payload_length_ms)
payload_length_ms = 10;
diff --git a/modules/audio_coding/codecs/opus/opus_interface.h b/modules/audio_coding/codecs/opus/opus_interface.h
index 54ecadd..cf95a69 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.h
+++ b/modules/audio_coding/codecs/opus/opus_interface.h
@@ -328,7 +328,9 @@
*/
int16_t WebRtcOpus_SetForceChannels(OpusEncInst* inst, size_t num_channels);
-int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, size_t channels);
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst,
+ size_t channels,
+ int sample_rate_hz);
/****************************************************************************
* WebRtcOpus_MultistreamDecoderCreate(...)
@@ -488,13 +490,15 @@
* Input:
* - payload : Encoded data pointer
* - payload_length_bytes : Bytes of encoded data
+ * - sample_rate_hz : Sample rate of output audio
*
* Return value : >0 - The duration of the FEC data in the
* packet in samples per channel.
* 0 - No FEC data in the packet.
*/
int WebRtcOpus_FecDurationEst(const uint8_t* payload,
- size_t payload_length_bytes);
+ size_t payload_length_bytes,
+ int sample_rate_hz);
/****************************************************************************
* WebRtcOpus_PacketHasFec(...)
diff --git a/modules/audio_coding/codecs/opus/opus_speed_test.cc b/modules/audio_coding/codecs/opus/opus_speed_test.cc
index 52cb14b..4477e8a 100644
--- a/modules/audio_coding/codecs/opus/opus_speed_test.cc
+++ b/modules/audio_coding/codecs/opus/opus_speed_test.cc
@@ -47,7 +47,7 @@
int app = channels_ == 1 ? 0 : 1;
/* Create encoder memory. */
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app, 48000));
- EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
+ EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_, 48000));
/* Set bitrate. */
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_));
}
diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc
index d0b240d..8a5bb6a 100644
--- a/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -67,9 +67,11 @@
void CreateSingleOrMultiStreamDecoder(WebRtcOpusDecInst** opus_decoder,
int channels,
- bool use_multistream) {
+ bool use_multistream,
+ int decoder_sample_rate_hz) {
EXPECT_TRUE(channels == 1 || channels == 2 || use_multistream);
if (use_multistream) {
+ EXPECT_EQ(decoder_sample_rate_hz, 48000);
if (channels == 1) {
EXPECT_EQ(0, WebRtcOpus_MultistreamDecoderCreate(
opus_decoder, channels, kMonoTotalStreams,
@@ -86,10 +88,16 @@
EXPECT_TRUE(false) << channels;
}
} else {
- EXPECT_EQ(0, WebRtcOpus_DecoderCreate(opus_decoder, channels));
+ EXPECT_EQ(0, WebRtcOpus_DecoderCreate(opus_decoder, channels,
+ decoder_sample_rate_hz));
}
}
+int SamplesPerChannel(int sample_rate_hz, int duration_ms) {
+ const int samples_per_ms = rtc::CheckedDivExact(sample_rate_hz, 1000);
+ return samples_per_ms * duration_ms;
+}
+
} // namespace
using test::AudioLoop;
@@ -99,16 +107,11 @@
// Maximum number of bytes in output bitstream.
const size_t kMaxBytes = 2000;
-// Sample rate of Opus.
-const size_t kOpusDecodeRateKhz = 48;
-// Number of samples-per-channel in a 20 ms frame, sampled at 48 kHz.
-const size_t kOpus20msFrameDecodeSamples = kOpusDecodeRateKhz * 20;
-// Number of samples-per-channel in a 10 ms frame, sampled at 48 kHz.
-const size_t kOpus10msFrameDecodeSamples = kOpusDecodeRateKhz * 10;
-class OpusTest : public TestWithParam<::testing::tuple<int, int, bool, int>> {
+class OpusTest
+ : public TestWithParam<::testing::tuple<size_t, int, bool, int, int>> {
protected:
- OpusTest();
+ OpusTest() = default;
void TestDtxEffect(bool dtx, int block_length_ms);
@@ -135,26 +138,35 @@
size_t channels,
uint16_t bound) const;
- WebRtcOpusEncInst* opus_encoder_;
- WebRtcOpusDecInst* opus_decoder_;
-
+ WebRtcOpusEncInst* opus_encoder_ = nullptr;
+ WebRtcOpusDecInst* opus_decoder_ = nullptr;
AudioLoop speech_data_;
uint8_t bitstream_[kMaxBytes];
- size_t encoded_bytes_;
- const size_t channels_;
- const int application_;
- const bool use_multistream_;
- const int encoder_sample_rate_hz_;
+ size_t encoded_bytes_ = 0;
+ const size_t channels_{std::get<0>(GetParam())};
+ const int application_{std::get<1>(GetParam())};
+ const bool use_multistream_{std::get<2>(GetParam())};
+ const int encoder_sample_rate_hz_{std::get<3>(GetParam())};
+ const int decoder_sample_rate_hz_{std::get<4>(GetParam())};
};
-OpusTest::OpusTest()
- : opus_encoder_(NULL),
- opus_decoder_(NULL),
- encoded_bytes_(0),
- channels_(static_cast<size_t>(::testing::get<0>(GetParam()))),
- application_(::testing::get<1>(GetParam())),
- use_multistream_(::testing::get<2>(GetParam())),
- encoder_sample_rate_hz_(::testing::get<3>(GetParam())) {}
+// Singlestream: Try all combinations.
+INSTANTIATE_TEST_SUITE_P(Singlestream,
+ OpusTest,
+ testing::Combine(testing::Values(1, 2),
+ testing::Values(0, 1),
+ testing::Values(false),
+ testing::Values(16000, 48000),
+ testing::Values(16000, 48000)));
+
+// Multistream: Some representative cases (only 48 kHz for now).
+INSTANTIATE_TEST_SUITE_P(
+ Multistream,
+ OpusTest,
+ testing::Values(std::make_tuple(1, 0, true, 48000, 48000),
+ std::make_tuple(2, 1, true, 48000, 48000),
+ std::make_tuple(4, 0, true, 48000, 48000),
+ std::make_tuple(4, 1, true, 48000, 48000)));
void OpusTest::PrepareSpeechData(int block_length_ms, int loop_length_ms) {
std::map<int, std::string> channel_to_basename = {
@@ -220,12 +232,14 @@
PrepareSpeechData(block_length_ms, 2000);
const size_t input_samples =
rtc::CheckedDivExact(encoder_sample_rate_hz_, 1000) * block_length_ms;
- const size_t output_samples = kOpusDecodeRateKhz * block_length_ms;
+ const size_t output_samples =
+ rtc::CheckedDivExact(decoder_sample_rate_hz_, 1000) * block_length_ms;
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
- CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_);
+ CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+ decoder_sample_rate_hz_);
// Set bitrate.
EXPECT_EQ(
@@ -385,7 +399,8 @@
// Test if CBR does what we expect.
void OpusTest::TestCbrEffect(bool cbr, int block_length_ms) {
PrepareSpeechData(block_length_ms, 2000);
- const size_t output_samples = kOpusDecodeRateKhz * block_length_ms;
+ const size_t output_samples =
+ rtc::CheckedDivExact(decoder_sample_rate_hz_, 1000) * block_length_ms;
int32_t max_pkt_size_diff = 0;
int32_t prev_pkt_size = 0;
@@ -393,7 +408,8 @@
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
- CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_);
+ CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+ decoder_sample_rate_hz_);
// Set bitrate.
EXPECT_EQ(
@@ -443,9 +459,11 @@
// Invalid sample rate.
EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 1, 0, 12345));
- EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(NULL, 1));
+ EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(NULL, 1, 48000));
// Invalid channel number.
- EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 257));
+ EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 257, 48000));
+ // Invalid sample rate.
+ EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 1, 12345));
}
// Test failing Free.
@@ -459,7 +477,8 @@
TEST_P(OpusTest, OpusCreateFree) {
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
- CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_);
+ CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+ decoder_sample_rate_hz_);
EXPECT_TRUE(opus_encoder_ != NULL);
EXPECT_TRUE(opus_decoder_ != NULL);
// Free encoder and decoder memory.
@@ -478,7 +497,8 @@
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
- CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_);
+ CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+ decoder_sample_rate_hz_);
// Set bitrate.
EXPECT_EQ(
@@ -495,12 +515,13 @@
// Encode & decode.
int16_t audio_type;
+ const int decode_samples_per_channel =
+ SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
int16_t* output_data_decode =
- new int16_t[kOpus20msFrameDecodeSamples * channels_];
- EXPECT_EQ(kOpus20msFrameDecodeSamples,
- static_cast<size_t>(
- EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
- opus_decoder_, output_data_decode, &audio_type)));
+ new int16_t[decode_samples_per_channel * channels_];
+ EXPECT_EQ(decode_samples_per_channel,
+ EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+ opus_decoder_, output_data_decode, &audio_type));
// Free memory.
delete[] output_data_decode;
@@ -549,8 +570,10 @@
PrepareSpeechData(20, 20);
int16_t audio_type;
+ const int decode_samples_per_channel =
+ SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
std::unique_ptr<int16_t[]> output_data_decode(
- new int16_t[kOpus20msFrameDecodeSamples * channels_]());
+ new int16_t[decode_samples_per_channel * channels_]());
// Test without creating encoder memory.
EXPECT_EQ(-1,
@@ -560,7 +583,8 @@
// Create encoder memory, try with different bandwidths.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
- CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_);
+ CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+ decoder_sample_rate_hz_);
EXPECT_EQ(-1, WebRtcOpus_SetBandwidth(opus_encoder_,
OPUS_BANDWIDTH_NARROWBAND - 1));
@@ -618,23 +642,24 @@
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
- CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_);
+ CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+ decoder_sample_rate_hz_);
// Encode & decode.
int16_t audio_type;
+ const int decode_samples_per_channel =
+ SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
int16_t* output_data_decode =
- new int16_t[kOpus20msFrameDecodeSamples * channels_];
- EXPECT_EQ(kOpus20msFrameDecodeSamples,
- static_cast<size_t>(
- EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
- opus_decoder_, output_data_decode, &audio_type)));
+ new int16_t[decode_samples_per_channel * channels_];
+ EXPECT_EQ(decode_samples_per_channel,
+ EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+ opus_decoder_, output_data_decode, &audio_type));
WebRtcOpus_DecoderInit(opus_decoder_);
- EXPECT_EQ(kOpus20msFrameDecodeSamples,
- static_cast<size_t>(
- WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
- output_data_decode, &audio_type)));
+ EXPECT_EQ(decode_samples_per_channel,
+ WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
+ output_data_decode, &audio_type));
// Free memory.
delete[] output_data_decode;
@@ -762,7 +787,8 @@
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
- CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_);
+ CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+ decoder_sample_rate_hz_);
// Set bitrate.
EXPECT_EQ(
@@ -773,18 +799,18 @@
// Encode & decode.
int16_t audio_type;
+ const int decode_samples_per_channel =
+ SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
int16_t* output_data_decode =
- new int16_t[kOpus20msFrameDecodeSamples * channels_];
- EXPECT_EQ(kOpus20msFrameDecodeSamples,
- static_cast<size_t>(
- EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
- opus_decoder_, output_data_decode, &audio_type)));
+ new int16_t[decode_samples_per_channel * channels_];
+ EXPECT_EQ(decode_samples_per_channel,
+ EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+ opus_decoder_, output_data_decode, &audio_type));
// Call decoder PLC.
- int16_t* plc_buffer = new int16_t[kOpus20msFrameDecodeSamples * channels_];
- EXPECT_EQ(
- kOpus20msFrameDecodeSamples,
- static_cast<size_t>(WebRtcOpus_DecodePlc(opus_decoder_, plc_buffer, 1)));
+ int16_t* plc_buffer = new int16_t[decode_samples_per_channel * channels_];
+ EXPECT_EQ(decode_samples_per_channel,
+ WebRtcOpus_DecodePlc(opus_decoder_, plc_buffer, 1));
// Free memory.
delete[] plc_buffer;
@@ -800,7 +826,8 @@
// Create.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
- CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_);
+ CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+ decoder_sample_rate_hz_);
// 10 ms. We use only first 10 ms of a 20 ms block.
auto speech_block = speech_data_.GetNextBlock();
@@ -809,10 +836,9 @@
rtc::CheckedDivExact(speech_block.size(), 2 * channels_), kMaxBytes,
bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
- EXPECT_EQ(
- kOpus10msFrameDecodeSamples,
- static_cast<size_t>(WebRtcOpus_DurationEst(
- opus_decoder_, bitstream_, static_cast<size_t>(encoded_bytes_int))));
+ EXPECT_EQ(SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/10),
+ WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
+ static_cast<size_t>(encoded_bytes_int)));
// 20 ms
speech_block = speech_data_.GetNextBlock();
@@ -821,10 +847,9 @@
rtc::CheckedDivExact(speech_block.size(), channels_),
kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
- EXPECT_EQ(
- kOpus20msFrameDecodeSamples,
- static_cast<size_t>(WebRtcOpus_DurationEst(
- opus_decoder_, bitstream_, static_cast<size_t>(encoded_bytes_int))));
+ EXPECT_EQ(SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20),
+ WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
+ static_cast<size_t>(encoded_bytes_int)));
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
@@ -846,7 +871,8 @@
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
ASSERT_NE(nullptr, opus_encoder_);
- CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_);
+ CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+ decoder_sample_rate_hz_);
ASSERT_NE(nullptr, opus_decoder_);
// Set bitrate.
@@ -858,8 +884,10 @@
// Encode & decode.
int16_t audio_type;
+ const int decode_samples_per_channel =
+ SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
std::unique_ptr<int16_t[]> output_data_decode(
- new int16_t[kPackets * kOpus20msFrameDecodeSamples * channels_]);
+ new int16_t[kPackets * decode_samples_per_channel * channels_]);
OpusRepacketizer* rp = opus_repacketizer_create();
size_t num_packets = 0;
@@ -887,11 +915,11 @@
encoded_bytes_ = opus_repacketizer_out(rp, bitstream_, kMaxBytes);
- EXPECT_EQ(kOpus20msFrameDecodeSamples * kPackets,
+ EXPECT_EQ(decode_samples_per_channel * kPackets,
static_cast<size_t>(WebRtcOpus_DurationEst(
opus_decoder_, bitstream_, encoded_bytes_)));
- EXPECT_EQ(kOpus20msFrameDecodeSamples * kPackets,
+ EXPECT_EQ(decode_samples_per_channel * kPackets,
static_cast<size_t>(
WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
output_data_decode.get(), &audio_type)));
@@ -902,21 +930,4 @@
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
-INSTANTIATE_TEST_SUITE_P(VariousMode,
- OpusTest,
- ::testing::ValuesIn({
- std::make_tuple(1, 0, false, 16000),
- std::make_tuple(1, 1, false, 16000),
- std::make_tuple(2, 0, false, 16000),
- std::make_tuple(2, 1, false, 16000),
- std::make_tuple(1, 0, false, 48000),
- std::make_tuple(1, 1, false, 48000),
- std::make_tuple(2, 0, false, 48000),
- std::make_tuple(2, 1, false, 48000),
- std::make_tuple(1, 0, true, 48000),
- std::make_tuple(2, 1, true, 48000),
- std::make_tuple(4, 0, true, 48000),
- std::make_tuple(4, 1, true, 48000),
- }));
-
} // namespace webrtc
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index 38d97a7..7f0bdd2 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -77,8 +77,8 @@
ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2, 1, 48000), -1);
// Create Opus decoders for mono and stereo for stand-alone testing of Opus.
- ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1), -1);
- ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2), -1);
+ ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1, 48000), -1);
+ ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2, 48000), -1);
WebRtcOpus_DecoderInit(opus_mono_decoder_);
WebRtcOpus_DecoderInit(opus_stereo_decoder_);