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Adding EncodedInfo struct to AudioEncoder::Encode

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This struct will be expanded in future changes.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7771 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
2014-12-01 14:44:50 +00:00
parent 20446e7e56
commit 1db20a4180
6 changed files with 15 additions and 12 deletions
Download Patch File Download Diff File Expand all files Collapse all files

10
webrtc/modules/audio_coding/codecs/audio_encoder.h
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@ -22,6 +22,10 @@ namespace webrtc {
// codec type must have an implementation of this class. // codec type must have an implementation of this class.
class AudioEncoder { class AudioEncoder {
public: public:
struct EncodedInfo {
uint32_t encoded_timestamp;
};
virtual ~AudioEncoder() {} virtual ~AudioEncoder() {}
// Accepts one 10 ms block of input audio (i.e., sample_rate_hz() / 100 * // Accepts one 10 ms block of input audio (i.e., sample_rate_hz() / 100 *
@ -37,7 +41,7 @@ class AudioEncoder {
size_t max_encoded_bytes, size_t max_encoded_bytes,
uint8_t* encoded, uint8_t* encoded,
size_t* encoded_bytes, size_t* encoded_bytes,
uint32_t* encoded_timestamp) { EncodedInfo* info) {
CHECK_EQ(num_samples_per_channel, CHECK_EQ(num_samples_per_channel,
static_cast<size_t>(sample_rate_hz() / 100)); static_cast<size_t>(sample_rate_hz() / 100));
bool ret = Encode(timestamp, bool ret = Encode(timestamp,
@ -45,7 +49,7 @@ class AudioEncoder {
max_encoded_bytes, max_encoded_bytes,
encoded, encoded,
encoded_bytes, encoded_bytes,
encoded_timestamp); info);
CHECK_LE(*encoded_bytes, max_encoded_bytes); CHECK_LE(*encoded_bytes, max_encoded_bytes);
return ret; return ret;
} }
@ -68,7 +72,7 @@ class AudioEncoder {
size_t max_encoded_bytes, size_t max_encoded_bytes,
uint8_t* encoded, uint8_t* encoded,
size_t* encoded_bytes, size_t* encoded_bytes,
uint32_t* encoded_timestamp) = 0; EncodedInfo* info) = 0;
}; };
} // namespace webrtc } // namespace webrtc

4
webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
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@ -57,7 +57,7 @@ bool AudioEncoderPcm::Encode(uint32_t timestamp,
size_t max_encoded_bytes, size_t max_encoded_bytes,
uint8_t* encoded, uint8_t* encoded,
size_t* encoded_bytes, size_t* encoded_bytes,
uint32_t* encoded_timestamp) { EncodedInfo* info) {
const int num_samples = sample_rate_hz() / 100 * num_channels(); const int num_samples = sample_rate_hz() / 100 * num_channels();
if (speech_buffer_.empty()) { if (speech_buffer_.empty()) {
first_timestamp_in_buffer_ = timestamp; first_timestamp_in_buffer_ = timestamp;
@ -72,7 +72,7 @@ bool AudioEncoderPcm::Encode(uint32_t timestamp,
CHECK_EQ(speech_buffer_.size(), static_cast<size_t>(full_frame_samples_)); CHECK_EQ(speech_buffer_.size(), static_cast<size_t>(full_frame_samples_));
int16_t ret = EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded); int16_t ret = EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
speech_buffer_.clear(); speech_buffer_.clear();
*encoded_timestamp = first_timestamp_in_buffer_; info->encoded_timestamp = first_timestamp_in_buffer_;
if (ret < 0) if (ret < 0)
return false; return false;
*encoded_bytes = static_cast<size_t>(ret); *encoded_bytes = static_cast<size_t>(ret);

2
webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
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@ -40,7 +40,7 @@ class AudioEncoderPcm : public AudioEncoder {
size_t max_encoded_bytes, size_t max_encoded_bytes,
uint8_t* encoded, uint8_t* encoded,
size_t* encoded_bytes, size_t* encoded_bytes,
uint32_t* encoded_timestamp) OVERRIDE; EncodedInfo* info) OVERRIDE;
virtual int16_t EncodeCall(const int16_t* audio, virtual int16_t EncodeCall(const int16_t* audio,
size_t input_len, size_t input_len,

4
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
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@ -76,7 +76,7 @@ bool AudioEncoderOpus::Encode(uint32_t timestamp,
size_t max_encoded_bytes, size_t max_encoded_bytes,
uint8_t* encoded, uint8_t* encoded,
size_t* encoded_bytes, size_t* encoded_bytes,
uint32_t* encoded_timestamp) { EncodedInfo* info) {
if (input_buffer_.empty()) if (input_buffer_.empty())
first_timestamp_in_buffer_ = timestamp; first_timestamp_in_buffer_ = timestamp;
input_buffer_.insert(input_buffer_.end(), audio, input_buffer_.insert(input_buffer_.end(), audio,
@ -97,7 +97,7 @@ bool AudioEncoderOpus::Encode(uint32_t timestamp,
if (r < 0) if (r < 0)
return false; return false;
*encoded_bytes = r; *encoded_bytes = r;
*encoded_timestamp = first_timestamp_in_buffer_; info->encoded_timestamp = first_timestamp_in_buffer_;
return true; return true;
} }

2
webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
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@ -40,7 +40,7 @@ class AudioEncoderOpus : public AudioEncoder {
size_t max_encoded_bytes, size_t max_encoded_bytes,
uint8_t* encoded, uint8_t* encoded,
size_t* encoded_bytes, size_t* encoded_bytes,
uint32_t* encoded_timestamp) OVERRIDE; EncodedInfo* info) OVERRIDE;
private: private:
const int num_10ms_frames_per_packet_; const int num_10ms_frames_per_packet_;

5
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
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@ -102,7 +102,6 @@ class AudioDecoderTest : public ::testing::Test {
data_length_(0), data_length_(0),
encoded_bytes_(0), encoded_bytes_(0),
channels_(1), channels_(1),
output_timestamp_(0),
decoder_(NULL) {} decoder_(NULL) {}
virtual ~AudioDecoderTest() {} virtual ~AudioDecoderTest() {}
@ -152,7 +151,7 @@ class AudioDecoderTest : public ::testing::Test {
EXPECT_TRUE(audio_encoder_->Encode( EXPECT_TRUE(audio_encoder_->Encode(
0, interleaved_input.get(), audio_encoder_->sample_rate_hz() / 100, 0, interleaved_input.get(), audio_encoder_->sample_rate_hz() / 100,
data_length_ * 2, output, &enc_len_bytes, &output_timestamp_)); data_length_ * 2, output, &enc_len_bytes, &encoded_info_));
} }
return static_cast<int>(enc_len_bytes); return static_cast<int>(enc_len_bytes);
} }
@ -263,7 +262,7 @@ class AudioDecoderTest : public ::testing::Test {
size_t data_length_; size_t data_length_;
size_t encoded_bytes_; size_t encoded_bytes_;
size_t channels_; size_t channels_;
uint32_t output_timestamp_; AudioEncoder::EncodedInfo encoded_info_;
AudioDecoder* decoder_; AudioDecoder* decoder_;
scoped_ptr<AudioEncoder> audio_encoder_; scoped_ptr<AudioEncoder> audio_encoder_;
}; };