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Set the maximum number of audio channels to 24

Browse source 
The number of audio channels can be configured in SDP, and can thus be
set to arbitrary values. However, the audio code has limitations that
prevent a high number of channels from working well in practice.

Bug: chromium:1265806
Change-Id: I6f6c3f68a3791bb189a614eece6bd0ed7874f252
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237807
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35359}
This commit is contained in:
Ivo Creusen 2021-11-16 15:11:28 +00:00 committed by WebRTC LUCI CQ
parent 584e3f9f8e
commit d823259c7f
12 changed files with 84 additions and 8 deletions
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3
api/audio_codecs/L16/audio_decoder_L16.h
View file

@ -29,7 +29,8 @@ struct RTC_EXPORT AudioDecoderL16 {
bool IsOk() const {
return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
num_channels >= 1;
(num_channels >= 1 &&
num_channels <= AudioDecoder::kMaxNumberOfChannels);
}
int sample_rate_hz = 8000;
int num_channels = 1;

4
api/audio_codecs/L16/audio_encoder_L16.h
View file

@ -29,7 +29,9 @@ struct RTC_EXPORT AudioEncoderL16 {
bool IsOk() const {
return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
num_channels >= 1 && frame_size_ms > 0 && frame_size_ms <= 120 &&
num_channels >= 1 &&
num_channels <= AudioEncoder::kMaxNumberOfChannels &&
frame_size_ms > 0 && frame_size_ms <= 120 &&
frame_size_ms % 10 == 0;
}
int sample_rate_hz = 8000;

1
api/audio_codecs/audio_decoder.cc
View file

@ -166,4 +166,5 @@ AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
}
}
constexpr int AudioDecoder::kMaxNumberOfChannels;
} // namespace webrtc

3
api/audio_codecs/audio_decoder.h
View file

@ -170,6 +170,9 @@ class AudioDecoder {
// during the lifetime of the decoder.
virtual size_t Channels() const = 0;
// The maximum number of audio channels supported by WebRTC decoders.
static constexpr int kMaxNumberOfChannels = 24;
protected:
static SpeechType ConvertSpeechType(int16_t type);

1
api/audio_codecs/audio_encoder.cc
View file

@ -110,4 +110,5 @@ ANAStats AudioEncoder::GetANAStats() const {
return ANAStats();
}
constexpr int AudioEncoder::kMaxNumberOfChannels;
} // namespace webrtc

3
api/audio_codecs/audio_encoder.h
View file

@ -246,6 +246,9 @@ class AudioEncoder {
virtual absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const = 0;
// The maximum number of audio channels supported by WebRTC encoders.
static constexpr int kMaxNumberOfChannels = 24;
protected:
// Subclasses implement this to perform the actual encoding. Called by
// Encode().

4
api/audio_codecs/g711/audio_decoder_g711.h
View file

@ -28,7 +28,9 @@ struct RTC_EXPORT AudioDecoderG711 {
struct Config {
enum class Type { kPcmU, kPcmA };
bool IsOk() const {
return (type == Type::kPcmU || type == Type::kPcmA) && num_channels >= 1;
return (type == Type::kPcmU || type == Type::kPcmA) &&
num_channels >= 1 &&
num_channels <= AudioDecoder::kMaxNumberOfChannels;
}
Type type;
int num_channels;

4
api/audio_codecs/g711/audio_encoder_g711.h
View file

@ -29,7 +29,9 @@ struct RTC_EXPORT AudioEncoderG711 {
enum class Type { kPcmU, kPcmA };
bool IsOk() const {
return (type == Type::kPcmU || type == Type::kPcmA) &&
frame_size_ms > 0 && frame_size_ms % 10 == 0 && num_channels >= 1;
frame_size_ms > 0 && frame_size_ms % 10 == 0 &&
num_channels >= 1 &&
num_channels <= AudioEncoder::kMaxNumberOfChannels;
}
Type type = Type::kPcmU;
int num_channels = 1;

3
api/audio_codecs/g722/audio_encoder_g722_config.h
View file

@ -15,7 +15,8 @@ namespace webrtc {
struct AudioEncoderG722Config {
bool IsOk() const {
return frame_size_ms > 0 && frame_size_ms % 10 == 0 && num_channels >= 1;
return frame_size_ms > 0 && frame_size_ms % 10 == 0 && num_channels >= 1 &&
num_channels <= AudioEncoder::kMaxNumberOfChannels;
}
int frame_size_ms = 20;
int num_channels = 1;

3
api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h
View file

@ -30,7 +30,8 @@ struct AudioDecoderMultiChannelOpusConfig {
std::vector<unsigned char> channel_mapping;
bool IsOk() const {
if (num_channels < 0 || num_streams < 0 || coupled_streams < 0) {
if (num_channels < 1 || num_channels > AudioDecoder::kMaxNumberOfChannels ||
num_streams < 0 || coupled_streams < 0) {
return false;
}
if (num_streams < coupled_streams) {

32
modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
View file

@ -104,9 +104,9 @@ TEST(AudioDecoderFactoryTest, CreateL16) {
rtc::scoped_refptr<AudioDecoderFactory> adf =
CreateBuiltinAudioDecoderFactory();
ASSERT_TRUE(adf);
// L16 supports any clock rate, any number of channels.
// L16 supports any clock rate and any number of channels up to 24.
const int clockrates[] = {8000, 16000, 32000, 48000};
const int num_channels[] = {1, 2, 3, 4711};
const int num_channels[] = {1, 2, 3, 24};
for (int clockrate : clockrates) {
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("l16", clockrate, 0),
absl::nullopt));
@ -117,6 +117,34 @@ TEST(AudioDecoderFactoryTest, CreateL16) {
}
}
// Tests that using more channels than the maximum does not work
TEST(AudioDecoderFactoryTest, MaxNrOfChannels) {
rtc::scoped_refptr<AudioDecoderFactory> adf =
CreateBuiltinAudioDecoderFactory();
std::vector<std::string> codecs = {
#ifdef WEBRTC_CODEC_OPUS
"opus",
#endif
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
"isac",
#endif
#ifdef WEBRTC_CODEC_ILBC
"ilbc",
#endif
"pcmu",
"pcma",
"l16",
"G722",
"G711",
};
for (auto codec : codecs) {
EXPECT_FALSE(adf->MakeAudioDecoder(
SdpAudioFormat(codec, 32000, AudioDecoder::kMaxNumberOfChannels + 1),
absl::nullopt));
}
}
TEST(AudioDecoderFactoryTest, CreateG722) {
rtc::scoped_refptr<AudioDecoderFactory> adf =
CreateBuiltinAudioDecoderFactory();

31
modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
View file

@ -144,4 +144,35 @@ TEST(BuiltinAudioEncoderFactoryTest, SupportsTheExpectedFormats) {
ASSERT_THAT(supported_formats, ElementsAreArray(expected_formats));
}
// Tests that using more channels than the maximum does not work.
TEST(BuiltinAudioEncoderFactoryTest, MaxNrOfChannels) {
rtc::scoped_refptr<AudioEncoderFactory> aef =
CreateBuiltinAudioEncoderFactory();
std::vector<std::string> codecs = {
#ifdef WEBRTC_CODEC_OPUS
"opus",
#endif
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
"isac",
#endif
#ifdef WEBRTC_CODEC_ILBC
"ilbc",
#endif
"pcmu",
"pcma",
"l16",
"G722",
"G711",
};
for (auto codec : codecs) {
EXPECT_FALSE(aef->MakeAudioEncoder(
/*payload_type=*/111,
/*format=*/
SdpAudioFormat(codec, 32000, AudioEncoder::kMaxNumberOfChannels + 1),
/*codec_pair_id=*/absl::nullopt));
}
}
} // namespace webrtc