diff options
Diffstat (limited to 'webrtc/modules/audio_coding/neteq/include/neteq.h')
| -rw-r--r-- | webrtc/modules/audio_coding/neteq/include/neteq.h | 27 |
1 files changed, 20 insertions, 7 deletions
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h index 205a0dfe80..1322223970 100644 --- a/webrtc/modules/audio_coding/neteq/include/neteq.h +++ b/webrtc/modules/audio_coding/neteq/include/neteq.h @@ -81,6 +81,7 @@ class NetEq { Config() : sample_rate_hz(16000), enable_audio_classifier(false), + enable_post_decode_vad(false), max_packets_in_buffer(50), // |max_delay_ms| has the same effect as calling SetMaximumDelay(). max_delay_ms(2000), @@ -92,6 +93,7 @@ class NetEq { int sample_rate_hz; // Initial value. Will change with input data. bool enable_audio_classifier; + bool enable_post_decode_vad; size_t max_packets_in_buffer; int max_delay_ms; BackgroundNoiseMode background_noise_mode; @@ -145,8 +147,7 @@ class NetEq { // the same tick rate as the RTP timestamp of the current payload. // Returns 0 on success, -1 on failure. virtual int InsertPacket(const WebRtcRTPHeader& rtp_header, - const uint8_t* payload, - size_t length_bytes, + rtc::ArrayView<const uint8_t> payload, uint32_t receive_timestamp) = 0; // Inserts a sync-packet into packet queue. Sync-packets are decoded to @@ -170,20 +171,27 @@ class NetEq { // The speech type is written to |type|, if |type| is not NULL. // Returns kOK on success, or kFail in case of an error. virtual int GetAudio(size_t max_length, int16_t* output_audio, - size_t* samples_per_channel, int* num_channels, + size_t* samples_per_channel, size_t* num_channels, NetEqOutputType* type) = 0; - // Associates |rtp_payload_type| with |codec| and stores the information in - // the codec database. Returns 0 on success, -1 on failure. + // Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the + // information in the codec database. Returns 0 on success, -1 on failure. + // The name is only used to provide information back to the caller about the + // decoders. Hence, the name is arbitrary, and may be empty. virtual int RegisterPayloadType(NetEqDecoder codec, + const std::string& codec_name, uint8_t rtp_payload_type) = 0; // Provides an externally created decoder object |decoder| to insert in the // decoder database. The decoder implements a decoder of type |codec| and - // associates it with |rtp_payload_type|. The decoder will produce samples - // at the rate |sample_rate_hz|. Returns kOK on success, kFail on failure. + // associates it with |rtp_payload_type| and |codec_name|. The decoder will + // produce samples at the rate |sample_rate_hz|. Returns kOK on success, kFail + // on failure. + // The name is only used to provide information back to the caller about the + // decoders. Hence, the name is arbitrary, and may be empty. virtual int RegisterExternalDecoder(AudioDecoder* decoder, NetEqDecoder codec, + const std::string& codec_name, uint8_t rtp_payload_type, int sample_rate_hz) = 0; @@ -250,6 +258,11 @@ class NetEq { // Returns true if the RTP timestamp is valid, otherwise false. virtual bool GetPlayoutTimestamp(uint32_t* timestamp) = 0; + // Returns the sample rate in Hz of the audio produced in the last GetAudio + // call. If GetAudio has not been called yet, the configured sample rate + // (Config::sample_rate_hz) is returned. + virtual int last_output_sample_rate_hz() const = 0; + // Not implemented. virtual int SetTargetNumberOfChannels() = 0; |