Merge "ASoC: msm: qdsp6v2: fix incomplete playback issue for non-gapless formats" into msm-4.8 (6a273ca0) · Commits · e / devices / android_kernel_fairphone_FP3

sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c

+34 −6

Original line number	Diff line number	Diff line
		@@ -604,6 +604,19 @@ static void compr_event_handler(uint32_t opcode,
		}
		}

		static int msm_compr_get_partial_drain_delay(int frame_sz, int sample_rate)
		{
		int delay_time_ms = 0;

		delay_time_ms = ((DSP_NUM_OUTPUT_FRAME_BUFFERED * frame_sz * 1000) /
		sample_rate) + DSP_PP_BUFFERING_IN_MSEC;
		delay_time_ms = delay_time_ms > PARTIAL_DRAIN_ACK_EARLY_BY_MSEC ?
		delay_time_ms - PARTIAL_DRAIN_ACK_EARLY_BY_MSEC : 0;

		pr_debug("%s: partial drain delay %d\n", __func__, delay_time_ms);
		return delay_time_ms;
		}

		static void populate_codec_list(struct msm_compr_audio *prtd)
		{
		pr_debug("%s\n", __func__);
		@@ -1225,8 +1238,9 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
		{
		struct snd_compr_runtime *runtime = cstream->runtime;
		struct msm_compr_audio *prtd = runtime->private_data;
		int ret = 0, frame_sz = 0, delay_time_ms = 0;
		int ret = 0, frame_sz = 0;
		int i, num_rates;
		bool is_format_gapless = false;

		pr_debug("%s\n", __func__);

		@@ -1265,6 +1279,7 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
		pr_debug("SND_AUDIOCODEC_MP3\n");
		prtd->codec = FORMAT_MP3;
		frame_sz = MP3_OUTPUT_FRAME_SZ;
		is_format_gapless = true;
		break;
		}

		@@ -1272,6 +1287,7 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
		pr_debug("SND_AUDIOCODEC_AAC\n");
		prtd->codec = FORMAT_MPEG4_AAC;
		frame_sz = AAC_OUTPUT_FRAME_SZ;
		is_format_gapless = true;
		break;
		}

		@@ -1279,6 +1295,7 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
		pr_debug("SND_AUDIOCODEC_AC3\n");
		prtd->codec = FORMAT_AC3;
		frame_sz = AC3_OUTPUT_FRAME_SZ;
		is_format_gapless = true;
		break;
		}

		@@ -1286,6 +1303,7 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
		pr_debug("SND_AUDIOCODEC_EAC3\n");
		prtd->codec = FORMAT_EAC3;
		frame_sz = EAC3_OUTPUT_FRAME_SZ;
		is_format_gapless = true;
		break;
		}

		@@ -1310,6 +1328,13 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
		case SND_AUDIOCODEC_FLAC: {
		pr_debug("%s: SND_AUDIOCODEC_FLAC\n", __func__);
		prtd->codec = FORMAT_FLAC;
		/*
		* DSP bufferring is based on blk size,
		* consider mininum buffering to rule out any false wait
		*/
		frame_sz =
		prtd->codec_param.codec.options.flac_dec.min_blk_size;
		is_format_gapless = true;
		break;
		}

		@@ -1336,11 +1361,11 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
		return -EINVAL;
		}

		delay_time_ms = ((DSP_NUM_OUTPUT_FRAME_BUFFERED * frame_sz * 1000) /
		prtd->sample_rate) + DSP_PP_BUFFERING_IN_MSEC;
		delay_time_ms = delay_time_ms > PARTIAL_DRAIN_ACK_EARLY_BY_MSEC ?
		delay_time_ms - PARTIAL_DRAIN_ACK_EARLY_BY_MSEC : 0;
		prtd->partial_drain_delay = delay_time_ms;
		if (!is_format_gapless)
		prtd->gapless_state.use_dsp_gapless_mode = false;

		prtd->partial_drain_delay =
		msm_compr_get_partial_drain_delay(frame_sz, prtd->sample_rate);

		memcpy(&prtd->codec_param, params, sizeof(struct snd_compr_params));

		@@ -2168,6 +2193,9 @@ static int msm_compr_set_metadata(struct snd_compr_stream *cstream,
		if (ret < 0)
		pr_err("%s: CMD Format block failed ret %d\n",
		__func__, ret);

		prtd->partial_drain_delay = msm_compr_get_partial_drain_delay(
		flac_cfg.min_blk_size, prtd->sample_rate);
		}
		return 0;
		}