SquashSRS4: Support RTC2RTMP.

trunk/src/app/srs_app_rtc_codec.hpp

@@ -26,6 +26,8 @@
 
 #include <srs_core.hpp>
 
+#include <srs_kernel_codec.hpp>
+
 #include <string>
 
 #ifdef __cplusplus
@@ -39,98 +41,59 @@ extern "C" {
 #include <libavutil/channel_layout.h>
 #include <libavutil/samplefmt.h>
 #include <libswresample/swresample.h>
+#include <libavutil/audio_fifo.h>
 
 #ifdef __cplusplus
 }
 #endif
 
-class SrsSample;
-
-class SrsAudioDecoder
+class SrsAudioTranscoder
 {
 private:
-    AVFrame* frame_;
-    AVPacket* packet_;
-    AVCodecContext* codec_ctx_;
-    SrsAudioCodecId codec_id_;
-public:
-    //Only support "aac","opus"
-    SrsAudioDecoder(SrsAudioCodecId codec);
-    virtual ~SrsAudioDecoder();
-    srs_error_t initialize();
-    virtual srs_error_t decode(SrsSample *pkt, char *buf, int &size);
-    AVCodecContext* codec_ctx();
-};
+    AVCodecContext *dec_;
+    AVFrame *dec_frame_;
+    AVPacket *dec_packet_;
 
-class SrsAudioEncoder
-{
+    AVCodecContext *enc_;
+    AVFrame *enc_frame_;
+    AVPacket *enc_packet_;
+
+    SwrContext *swr_;
+    //buffer for swr out put
+    uint8_t **swr_data_;
+    AVAudioFifo *fifo_;
+
+    int64_t new_pkt_pts_;
+    int64_t next_out_pts_;
+public:
+    SrsAudioTranscoder();
+    virtual ~SrsAudioTranscoder();
+public:
+    // Initialize the transcoder, transcode from codec as to codec.
+    // The channels specifies the number of output channels for encoder, for example, 2.
+    // The sample_rate specifies the sample rate of encoder, for example, 48000.
+    // The bit_rate specifies the bitrate of encoder, for example, 48000.
+    srs_error_t initialize(SrsAudioCodecId from, SrsAudioCodecId to, int channels, int sample_rate, int bit_rate);
+    // Transcode the input audio frame in, as output audio frames outs.
+    virtual srs_error_t transcode(SrsAudioFrame* in, std::vector<SrsAudioFrame*>& outs);
+    // Free the generated audio frames by transcode.
+    void free_frames(std::vector<SrsAudioFrame*>& frames);
+public:
+    // Get the aac codec header, for example, FLV sequence header.
+    // @remark User should never free the data, it's managed by this transcoder.
+    void aac_codec_header(uint8_t** data, int* len);
 private:
-	int channels_;
-	int sampling_rate_;
-    AVCodecContext* codec_ctx_;
-    SrsAudioCodecId codec_id_;
-    int want_bytes_;
-    AVFrame* frame_;
-public:
-    //Only support "aac","opus"
-    SrsAudioEncoder(SrsAudioCodecId codec, int samplerate, int channelsy);
-    virtual ~SrsAudioEncoder();
-    srs_error_t initialize();
-    //The encoder wanted bytes to call encode, if > 0, caller must feed the same bytes
-    //Call after initialize successed
-    int want_bytes();
-    virtual srs_error_t encode(SrsSample *frame, char *buf, int &size);
-    AVCodecContext* codec_ctx();
-};
+    srs_error_t init_dec(SrsAudioCodecId from);
+    srs_error_t init_enc(SrsAudioCodecId to, int channels, int samplerate, int bit_rate);
+    srs_error_t init_swr(AVCodecContext* decoder);
+    srs_error_t init_fifo();
 
-class SrsAudioResample
-{
-private:
-    int src_rate_;
-    int src_ch_layout_;
-    int src_nb_channels_;
-    enum AVSampleFormat src_sample_fmt_;
-    int src_linesize_;
-    int src_nb_samples_;
-    uint8_t **src_data_;
+    srs_error_t decode_and_resample(SrsAudioFrame* pkt);
+    srs_error_t encode(std::vector<SrsAudioFrame*> &pkts);
 
-    int dst_rate_;
-    int dst_ch_layout_;
-    int dst_nb_channels_;
-    enum AVSampleFormat dst_sample_fmt_;
-    int dst_linesize_;
-    int dst_nb_samples_;
-    uint8_t **dst_data_;
-
-    int max_dst_nb_samples_;
-    struct SwrContext *swr_ctx_;
-public:
-    SrsAudioResample(int src_rate, int src_layout, enum AVSampleFormat src_fmt,
-        int src_nb, int dst_rate, int dst_layout, enum AVSampleFormat dst_fmt);
-    virtual ~SrsAudioResample();
-    srs_error_t initialize();
-    virtual srs_error_t resample(SrsSample *pcm, char *buf, int &size);
-};
-
-// TODO: FIXME: Rename to Transcoder.
-class SrsAudioRecode
-{
-private:
-    SrsAudioDecoder *dec_;
-    SrsAudioEncoder *enc_;
-    SrsAudioResample *resample_;
-    int dst_channels_;
-    int dst_samplerate_;
-    int size_;
-    char *data_;
-    SrsAudioCodecId src_codec_;
-    SrsAudioCodecId dst_codec_;
-    int enc_want_bytes_;
-public:
-    SrsAudioRecode(SrsAudioCodecId src_codec, SrsAudioCodecId dst_codec,int channels, int samplerate);
-    virtual ~SrsAudioRecode();
-    srs_error_t initialize();
-    virtual srs_error_t transcode(SrsSample *pkt, char **buf, int *buf_len, int &n);
+    srs_error_t add_samples_to_fifo(uint8_t** samples, int frame_size);
+    void free_swr_samples();
 };
 
 #endif /* SRS_APP_AUDIO_RECODE_HPP */
+