Adding a payload type to AudioEncoder objects
The type is set in the Config struct and is provided in the EncodedInfo
output struct from each Encode() call. The audio_decoder_unittest is
updated to verify correct propagation of the payload type.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7780 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index 6f68d5c..c0790a2 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -24,6 +24,7 @@
public:
struct EncodedInfo {
uint32_t encoded_timestamp;
+ int payload_type;
};
virtual ~AudioEncoder() {}
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
index ee8dfeb..7f0cc0d 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -29,6 +29,7 @@
AudioEncoderPcm::AudioEncoderPcm(const Config& config)
: num_channels_(config.num_channels),
+ payload_type_(config.payload_type),
num_10ms_frames_per_packet_(config.frame_size_ms / 10),
full_frame_samples_(NumSamplesPerFrame(num_channels_,
config.frame_size_ms,
@@ -73,6 +74,7 @@
int16_t ret = EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
speech_buffer_.clear();
info->encoded_timestamp = first_timestamp_in_buffer_;
+ info->payload_type = payload_type_;
if (ret < 0)
return false;
*encoded_bytes = static_cast<size_t>(ret);
diff --git a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
index b3866f0..71a72c8 100644
--- a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
+++ b/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
@@ -20,10 +20,14 @@
class AudioEncoderPcm : public AudioEncoder {
public:
struct Config {
- Config() : frame_size_ms(20), num_channels(1) {}
-
+ public:
int frame_size_ms;
int num_channels;
+ int payload_type;
+
+ protected:
+ explicit Config(int pt)
+ : frame_size_ms(20), num_channels(1), payload_type(pt) {}
};
explicit AudioEncoderPcm(const Config& config);
@@ -49,6 +53,7 @@
private:
static const int kSampleRateHz = 8000;
const int num_channels_;
+ const int payload_type_;
const int num_10ms_frames_per_packet_;
const int16_t full_frame_samples_;
std::vector<int16_t> speech_buffer_;
@@ -57,6 +62,10 @@
class AudioEncoderPcmA : public AudioEncoderPcm {
public:
+ struct Config : public AudioEncoderPcm::Config {
+ Config() : AudioEncoderPcm::Config(8) {}
+ };
+
explicit AudioEncoderPcmA(const Config& config) : AudioEncoderPcm(config) {}
protected:
@@ -67,6 +76,10 @@
class AudioEncoderPcmU : public AudioEncoderPcm {
public:
+ struct Config : public AudioEncoderPcm::Config {
+ Config() : AudioEncoderPcm::Config(0) {}
+ };
+
explicit AudioEncoderPcmU(const Config& config) : AudioEncoderPcm(config) {}
protected:
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
index ccc6c77..d64f830 100644
--- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -33,6 +33,7 @@
AudioEncoderG722::AudioEncoderG722(const Config& config)
: num_channels_(config.num_channels),
+ payload_type_(config.payload_type),
num_10ms_frames_per_packet_(config.frame_size_ms / 10),
num_10ms_frames_buffered_(0),
first_timestamp_in_buffer_(0),
@@ -113,6 +114,7 @@
}
*encoded_bytes = samples_per_channel / 2 * num_channels_;
info->encoded_timestamp = first_timestamp_in_buffer_;
+ info->payload_type = payload_type_;
return true;
}
diff --git a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
index 8e4de22..51cba7b 100644
--- a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
+++ b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
@@ -53,6 +53,7 @@
};
const int num_channels_;
+ const int payload_type_;
const int num_10ms_frames_per_packet_;
int num_10ms_frames_buffered_;
uint32_t first_timestamp_in_buffer_;
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index d7ca1a4..35261bc 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -36,7 +36,9 @@
} // namespace
-AudioEncoderOpus::Config::Config() : frame_size_ms(20), num_channels(1) {}
+AudioEncoderOpus::Config::Config()
+ : frame_size_ms(20), num_channels(1), payload_type(120) {
+}
bool AudioEncoderOpus::Config::IsOk() const {
if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
@@ -49,6 +51,7 @@
AudioEncoderOpus::AudioEncoderOpus(const Config& config)
: num_10ms_frames_per_packet_(DivExact(config.frame_size_ms, 10)),
num_channels_(config.num_channels),
+ payload_type_(config.payload_type),
samples_per_10ms_frame_(DivExact(kSampleRateHz, 100) * num_channels_) {
CHECK(config.IsOk());
input_buffer_.reserve(num_10ms_frames_per_packet_ * samples_per_10ms_frame_);
@@ -98,6 +101,7 @@
return false;
*encoded_bytes = r;
info->encoded_timestamp = first_timestamp_in_buffer_;
+ info->payload_type = payload_type_;
return true;
}
diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
index 2071c0f..3cbb25b 100644
--- a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
@@ -25,6 +25,7 @@
bool IsOk() const;
int frame_size_ms;
int num_channels;
+ int payload_type;
};
explicit AudioEncoderOpus(const Config& config);
@@ -45,6 +46,7 @@
private:
const int num_10ms_frames_per_packet_;
const int num_channels_;
+ const int payload_type_;
const int samples_per_10ms_frame_;
std::vector<int16_t> input_buffer_;
OpusEncInst* inst_;
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 191e81a..bbcf9ed 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -102,6 +102,7 @@
data_length_(0),
encoded_bytes_(0),
channels_(1),
+ payload_type_(17),
decoder_(NULL) {}
virtual ~AudioDecoderTest() {}
@@ -153,6 +154,7 @@
0, interleaved_input.get(), audio_encoder_->sample_rate_hz() / 100,
data_length_ * 2, output, &enc_len_bytes, &encoded_info_));
}
+ EXPECT_EQ(payload_type_, encoded_info_.payload_type);
return static_cast<int>(enc_len_bytes);
}
@@ -262,6 +264,7 @@
size_t data_length_;
size_t encoded_bytes_;
size_t channels_;
+ const int payload_type_;
AudioEncoder::EncodedInfo encoded_info_;
AudioDecoder* decoder_;
scoped_ptr<AudioEncoder> audio_encoder_;
@@ -275,6 +278,7 @@
decoder_ = new AudioDecoderPcmU;
AudioEncoderPcmU::Config config;
config.frame_size_ms = static_cast<int>(frame_size_ / 8);
+ config.payload_type = payload_type_;
audio_encoder_.reset(new AudioEncoderPcmU(config));
}
};
@@ -287,6 +291,7 @@
decoder_ = new AudioDecoderPcmA;
AudioEncoderPcmA::Config config;
config.frame_size_ms = static_cast<int>(frame_size_ / 8);
+ config.payload_type = payload_type_;
audio_encoder_.reset(new AudioEncoderPcmA(config));
}
};
@@ -485,6 +490,7 @@
assert(decoder_);
AudioEncoderG722::Config config;
config.frame_size_ms = 10;
+ config.payload_type = payload_type_;
config.num_channels = 1;
audio_encoder_.reset(new AudioEncoderG722(config));
}
@@ -501,6 +507,7 @@
assert(decoder_);
AudioEncoderG722::Config config;
config.frame_size_ms = 10;
+ config.payload_type = payload_type_;
config.num_channels = 2;
audio_encoder_.reset(new AudioEncoderG722(config));
}
@@ -586,6 +593,7 @@
decoder_ = new AudioDecoderOpus(1);
AudioEncoderOpus::Config config;
config.frame_size_ms = static_cast<int>(frame_size_) / 48;
+ config.payload_type = payload_type_;
audio_encoder_.reset(new AudioEncoderOpus(config));
}
};
@@ -599,6 +607,7 @@
AudioEncoderOpus::Config config;
config.frame_size_ms = static_cast<int>(frame_size_) / 48;
config.num_channels = 2;
+ config.payload_type = payload_type_;
audio_encoder_.reset(new AudioEncoderOpus(config));
}
};
